Analog PSTN - ISDN

Hello,
I have a FAX server which need to be connected with ISDN lines. And as the solution is in Algeria where there is no ISDN lines.
I think about using a router for conversion from analog to ISDN:
Server|---(ISDN)---|Router(FXO)|------|Analog
Is this possible ? or can I have another alternative for the solution?
Regards,
Omar

Hi Omar,
You can use an interface like T1 CAS or E1 R2 if the fax server supports it. This is going to be limited by the fax server, not the router.
And yes, the router can do network side of the PRI. Simply add 'isdn protocol-emulate network' under the 'Serial 0/0/0:23' interface.
hth,
nick

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Handle NOTIFY isdn message

In advance to reduce the time that a call is closed after the called hangup i need to instruct my gateways to close the call when isdn NOTIFY message is received. What is the settings needed to do this ?
I see in debugs that the pstn isdn switch send a NOTIFY message when the called subscriber hangup, 60 seconds later i receive the DISCONNECT message from switch.
In TCLHandler the event of NOTIFY is received as EV_CALL_SUSPEND_ACK but i cannot handle this with:
set fsm(CALLACTIVE,ev_suspend_ack) "act_Cleanup,CALLDISCONNECT"
Thanks

Were you able to resolve the above issue, as we are currently running into the same.
Whenever the called party hang's up we get the disconnect message after 30 secs.
Is there a solution to this problem, as generally here in the US its a FCC regulation so CO dont send Disconnect whenever the called party hangs up, instead they recv a SS7 suspend which invokes a suspend timer on the orig Switch and finally when the timer expires the call tear down happens.
We have spoken to the carrier, if they can send us a Q931 notify, or suspend so that we can act upon it.
current scenario
SS7---ISDN/H323----GK----CM--IP Phone
Thanks

[SOLVED] USB 3G no longer works after upgrade to GNOME 3.8

I did the 'big' update today, including upgrading to GNOME 3.8. Unfortunately, NetworkManager no longer picks up my 3G USB modem (which used to work fine before the update). Any pointers as to how I could debug this would be greatly appreciated.
wvdial has never really worked for me, but gnome networkmanager used to pick up my device and the connection, and now simply acts as if it's no longer present. Attempting to follow the instructions on the wiki to re-setup my USB stick (the instructions I followed in the first place) no longer works. Did today's update cause the same problem for anybody else?
**Update: After updating today to modemmanager 0.7.990-4 and running
sudo systemctl enable ModemManager
sudo systemctl start ModemManager
I have full connectivity again with the built-in DW5550 mobile broadband card in my Dell Precision M6600 (including showing signal strength in the GNOME 3.8 top bar, etc). Thanks all for the help, and (most importantly) to whoever fixed up modemmanager!
Last edited by dawid.loubser (2013-04-26 16:57:42)

wvdial might also be a workaround, if one does not like to downgrade. In my case, I had to unset my PIN on the SIM card first. One can do that using a mobile phone, for example.
Then
# pacman -S wvdial
in /etc/wvdial.conf, I put
[Dialer Defaults]
Init2 = ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0
Modem Type = Analog Modem
ISDN = 0
Stupid Mode = 1
Dial Attempts = 3
Init1 = ATZ
[Dialer huawei]
Modem = /dev/serial/by-id/usb-ZTE_Incorporated_1_1_Surf-stick_MF19001MOD010000-if02-port0
Baud = 115200
[Dialer voda]
Init3 = AT+CGDCONT=1,"IP","web.vodafone.de"
[Dialer dial]
Phone=*99**1#
Username = umts
Password = umts
You can find out your modem name by looking it up with
ls /dev/serial/by-id/
In my case there were names containing the strings if00, if01 and if02, and if02 turned out to be the correct one. This depends on hardware.
Also, in the "Init 3" line, you may have to replace "web.vodafone.de" according to your provider. The section "dial" seems to be quite universal from what I read on the web, so you may not have to change that.
then
# wvdial huawei voda dial
and hope for the best.
Disclaimer: I do not have a comprehensive understanding of what I did here, but it worked in my case and it might work for you, too.
I noticed that the Defaults section makes a difference, because the one I just edited in above avoids some delay in dialing.
Last edited by tobi-fankhaenel (2013-04-23 08:23:00)

NAC with security rtr

hello
we want to implement a NAC solution for people dialing from home to HO then going to internet via our internet router.
this router contains the security feature and is NAC enabled (we can see this from web interface)
however, one cisco partner suggests to use clean access server and not the security router.
is there any advantage of using clean access servers or limitation for security rtr.
note: we only need to check for windows updates and antivirus updates when computers access internet

Well, both NAC Framework (NAC on your router) and NAC Appliance (Clean Access Server) will work. You can dial via PSTN/ISDN or via VPN using Cisco VPN Client. Also, you can purchase NME-NAC-K9 module for your router and it will work like Clean Access Server.
To use NAC Framework you'll also need Cisco Secure Access Control Server (CS ACS) 4.0+ (4.1). This is commercial RADIUS server and isn't cheap.
Also, to check for antivirus updates your antivirus product must be supported by either NAC Framework or Appliance. For a list of supported products take a look at:
http://www.cisco.com/go/nac
http://www.cisco.com/web/partners/pr46/nac/partners.html (NAC Framework)
http://www.cisco.com/en/US/docs/security/nac/appliance/release_notes/416/416rn.html (NAC Appliance)
For NAC Framework you'll need to integrate vendor .dlls into the Cisco Trust Agent (for all of your antivirus vendors!), then distribute CTA to all user PCs using some out-of-band mechanism (not an easy task). CTA is a must for NAC Framework.
NAC Appliance automates this. This is a self-contained product (no .dlls). Clean Access Agent can check supported antivirus products by itself. It can be installed onto PCs via some out-of-band mechanism or downloaded from the Web Login page. Also, Java / ActiveX agent is supported and can check your PC for compliance as well.
Checking for Service Pack number isn't
difficult in both products. However, to check for Windows Hotfixes you'll have to create complex rules in NAC Framework. When a new hotfix is released by Microsoft you'll have to change your rules manually (not easy). NAC Appliance automates this. It can download rules from the Cisco website. But you'll have to buy tech support for this.
In general, configuring and maintaining NAC Framework is not an easy task. However, you can buy additional products, integrate them into the Framework and they will automate many things for you. This is not cheap and easy. NAC Appliance is self-contained. You'll not need anything else.
HTH

Cisco CME / 7925G: Leading zero stripped in Placed Calls list

Hi,
we are using a "0" as a prefix to dialout to the PSTN (ISDN PRI) on our CME router (2921).
The leading 0 will be stripped off on the dial peer via a voice translation rule:
voice translation-rule 11
rule 1 /^00\(.*\)/ /0\1/
voice translation-profile OUTGOING_TRANSLATION
translate called 11
Problem: When we dial an external number (including the leading 0) on the 7925G phone, it stores this dialed number
without the leading 0 in the "placed calls" list.
When you want to pick up and call a number later from that list, it won´t work, because the leading 0 is missing.
Does anyone know how to fix this?
Many thanks
Heinz

Hi Heinz,I don't know if this kind of manipulation is available in CME or not however as per your comments, you can give a try to following.As you are stripping leading '0' at dial-peer level, now while dialing the number from logs, you can again prefix '0' at ephone-dn level at incoming call leg. Use following translation rule in one of the phone and see if it works for you. This rule will work completely depends on number patterns dialed by your users. While dialing the number initially by prefixing '00', if there is a possibility to dial third digit as '0', this rule won't work.
rule 1 /^0\([^0]\)\(.*\)/ /00\1\2/Now if you initially dial 009925801882, you are stripping initial '0'. This won't match the number at ephone-dn level and only match at dial-peer level. In logs, you are seeing 09925801882. Now when you dial this number from logs viz 09925801882, translation profile at ephone-dn level will change it to 009925801882 and will again strip '0' at dial-peer level. ThanksVivek

Audio Streaming (MOH)

Hi, I need to steam audio (audio instead of call signalling/moh) from IP Gateway to IP Gateway (GK or SIP but NO PSTN/ISDN GW). Where can I find information /documents about scripting or streaming on cisco routers.
thx patricio

Hi,
Can you tell how you can get CM installed in VMWare ? Can I get instructions from you ?
Thanks !!
Danny

Remote support zone to customers

Hi,
We are looking to setup a remote support zone to different customers (more than 500) for our different software developers.
The type of connections to customers can be PSTN/ISDN dialup, PPTP and ipsec L2L connections.
We don't want that the connections can be initiated from within the local lan so we thought of having a DMZ with terminal server and workstations where the different teams can logon.
Different needs :
Terminal server & workstations (for debugging) in zone
One team can only access his customers from Terminal server and from his workstations.
AAA
We thought of using the new ISR routers for dialup and ipsec in combination with ACS.
For PPTP we still need a Windows server.
a few remarks : how is it possible to use ACS with auth-proxy if you let the users login to the same TS (so same source ip ?)
Can ACS be used together with the microsoft server for setting up PPTP connections ?
Any ideas about architecture and a secure zone are welcome.
Regards,
Kristine

As far as I know, you cannot use ACS with auth-proxy

FXS and FXO configuration and design help

hi all...
i am new to voip configuration... and i have to configure the voip for our branch office... we have 2610 with the 2FXS and 2FXO prot... now i have one pulblic IP address which is used by my local lan user...and we have configured NAT/PAT on our VPN concerntrator... now we are looking to establish the VOIP phone (analogphone with the help of FXO and FXS) we have ASTRIK server at our main office and i want to register my local office analog phone with server at our main office... now what kind of configuration we need on our 2610 in order to configure voip...
connectivity:
ADSL connection form ISP---VPN 3005---D-Link nonmanagable swithch---LAN
at present we have above connectivity and now i want to add my 2610 router with analogphone connected to it... how can i connect and how can i configure it...?
regards
Devang

you need to configure pots and voip dialPeers for the incoming/outgoing legs of the calls.
pots dialPeers will be used for the FXO/FXS ports. voip dialPeers will be used for connection to ASTERISK voip pbx.
you need voice port configuration for your FXS to connect to your analogPhone.
you need voice port configuration for your FXO to connect to analog pstn. (if you have any)
see these links for more info:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080147524.shtml
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801bc341.shtml
http://www.cisco.com/en/US/tech/tk652/tk653/tech_configuration_examples_list.html
there are plenty of examples throughout these for reference.

FXO and Telco CO Transfer?

We are planning an IPT installatio for a small branch office. We will have 4-6 analog PSTN lines, one line assigned to each phone. The telco is offering a transfer feature on the lines, which would allow the receptionist to transfer calls to the other lines. The main number would then be free to receive another call, instead of being tied up as it would be if the transfer was made internally. Is there somehow to support this feature with our Cisco IPT implementation? Thanks.

if you've got call manager express running then you can have it perform your transfer features.
are you using call manager express? if so, you can setup transfer statements in the router.
see this link for more info:
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapter09186a00804ad858.html

Router 2901 as VoIP gateway

Hello,
I'm trying to configure a CISCO 2901 Integrated Services Router with a Module VWIC3-1MFT-T1/E1 as a VoIP gateway.
I'd like to know if is it possible, and if there is a especific document in CISCO documentation explaining it.
I have the following escenary:
Inbound calls:
PSTN (ISDN) --> CISCO 2901 --> Asterisk (SIP PBX)
Outbound calls
Asterisk(SIP PBX) --> CISCO 2901 --> PSTN (ISDN)
I have found some documents which explains how to configure a SIP Trunk with registration
http://www.cisco.com/en/US/docs/ios-xml/ios/voice/cube_sipsip/configuration/xe-3s/asr1000/voi-sip-trunk-reg.html
And how to configure the VWIC3 module
http://www.cisco.com/en/US/docs/routers/access/interfaces/software/feature/guide/vd-t1e1_vwic3.html
I'm very new with CISCO and I don't know how to route calls from the module to the Ethernet interface, and if I need to do anything else.
Thanks in advance!
David Pinedo
Presence Technology

For ISDN you need to configure your T1/E1, plenty of material and also references/examples on CSC, a quick search will provide some basic config
http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucme/admin/configuration/guide/cmebasic.html
Search pri-group timeslots to get more examples.
Then you'll need to configure dial-peers for in/out calls, search cisco.com for dial-peer configuration and you'll get all the info you need on how to configure them.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk

Dialup scenario

Hi
I need some expert advice. I have been asked to provide a solution to the following.
20 remote sites with analog PSTN only require VPN access via the Internet to a central site.
the devices within the remote sites which require this access are Point of sale devices with no VPN software running on them. They will need to be attached to a device(s) which can seamlessly dial an ISP,recieve a dynamic IP address, establish an Internet connection then build a VPN tunnel to the central site and send the data across the tunnel.
Can anyone suggest a cisco device (805 maybe?)that can do this and perhaps provide me with configuration suggestions.
Thank you

A solution like this is deemed to deliver bad performance from day 1. There will simply not be enough bandwidth on a dialup line to accomodate a vpn connection, so be warned.
I assume that there will not be more than 1-2 users per location? In that case you will not need a Cisco router on the remote sites. Instead you should consider to implement the security on the client-PC's. Put an ASA firewall in the center and use the Cisco VPN client to connect when needed. It will make your project a lot cheaper too!
Regards,
Leo

E1 PRI Links Calculation

Dears,
We are moving from old telephony to IP Telephony, just we are in a design phase so need to worry for each and everything.
I want to know the below information as per the cisco standard,
we have 1000 no of users so how many E1 PRI we need
what should be the WAN bandwidth to avoid database replication between a publisher and subscriber through WAN, as cisco doc says the RTT should not be more than 80ms, so how much bandwith i shld ask from the ITSP.

PRI Info:
For sizing you will need the Erlang-B calculator http://www.erlang.com/calculator/
WAN Design Info:
The round-trip time (RTT) must not exceed the currently supported limit of 80 ms for clustering over the WAN. To calculate the necessary bandwidth for CTI traffic, use the formula in the section on Local Failover Deployment Model. Note that this bandwidth is in addition to the Intra-Cluster Communication Signaling (ICCS) bandwidth calculated as described in the section on Local Failover Deployment Model, as well as any bandwidth required for audio (RTP traffic).
For clustering over the WAN to be successful, you must carefully plan, design, and implement various characteristics of the WAN itself. The Intra-Cluster Communication Signaling (ICCS) between Unified CM servers consists of many traffic types. The ICCS traffic types are classified as either priority or best-effort. Priority ICCS traffic is marked with IP Precedence 3 (DSCP 24 or PHB CS3). The following design guidelines apply to the indicated WAN characteristics:
•Delay
The maximum one-way delay between any two Unified CM servers should not exceed 40 msec, or 80 msec round-trip time. Measuring the delay is covered in Delay Testing. Propagation delay between two sites introduces 6 microseconds per kilometer without any other network delays being considered. This equates to a theoretical maximum distance of approximately 3000 km for 20 ms delay or approximately 1860 miles. These distances are provided only as relative guidelines and in reality will be shorter due to other delay incurred within the network.
•Jitter
Jitter is the varying delay that packets incur through the network due to processing, queue, buffer, congestion, or path variation delay. Jitter for the IP Precedence 3 ICCS traffic must be minimized using Quality of Service (QoS) features.
•Packet loss and errors
The network should be engineered to provide sufficient prioritized bandwidth for all ICCS traffic, especially the priority ICCS traffic. Standard QoS mechanisms must be implemented to avoid congestion and packet loss. If packets are lost due to line errors or other "real world" conditions, the ICCS packet will be retransmitted because it uses the TCP protocol for reliable transmission. The retransmission might result in a call being delayed during setup, disconnect (teardown), or other supplementary services during the call. Some packet loss conditions could result in a lost call, but this scenario should be no more likely than errors occurring on a T1 or E1, which affect calls via a trunk to the PSTN/ISDN.
•Bandwidth
Provision the correct amount of bandwidth between each server for the expected call volume, type of devices, and number of devices. This bandwidth is in addition to any other bandwidth for other applications sharing the network, including voice and video traffic between the sites. The bandwidth provisioned must have QoS enabled to provide the prioritization and scheduling for the different classes of traffic. The general rule of thumb for bandwidth is to over-provision and under-subscribe.
•Quality of Service
The network infrastructure relies on QoS engineering to provide consistent and predictable end-to-end levels of service for traffic. Neither QoS nor bandwidth alone is the solution; rather, QoS-enabled bandwidth must be engineered into the network infrastructure.
Intra-Cluster Communications
In general, intra-cluster communications means all traffic between servers. There is also a real-time protocol called Intra-Cluster Communication Signaling (ICCS), which provides the communications with the Cisco CallManager Service process that is at the heart of the call processing in each server or node within the cluster.
The intra-cluster traffic between the servers consists of the following:
•Database traffic from the IBM Informix Dynamic Server (IDS) database that provides the main configuration information. The IDS traffic may be re-prioritized in line with Cisco QoS recommendations to a higher priority data service (for example, IP Precedence 1 if required by the particular business needs). An example of this is extensive use of Extension Mobility, which relies on IDS database configuration.
•ICCS real-time traffic, which consists of signaling, call admission control, and other information regarding calls as they are initiated and completed.
HTH
Regards,
Yosh

Problem with cisco series 800 router and SMTP

Hello,
we bought a 877 cisco router and have some problem with SMTP.
I try to read all forum and KB about but do not find any solution.
the problem is that when i try to send any email from client (windows mail) i receive a error 533: you need to supply the correct username and password.
when trying throught hmailserver i receive an email "undeliverable" with this error in body:
Error Type: SMTP
Remote server (62.149.128.202) issued an error.
hMailServer sent: EHLO Globalnet
Remote server replied: 502 unimplemented (#5.5.1)
receving email work correctly.
i'm already using smtp auth, and with my old router everything work fine.
so i beleive is a config problem, maybe with ESMTP (ehlo)?.
i attach my config:

Dear sirs
Thank you for answer so quickly
I download this document from Cisco âConfiguring the (Remote) Common Application Programming Interface for Cisco 800 Series Routerâ.
I have a Lan with Asterisk IP-PBX , the Cisco router have a BRI to public exchange (ISDN) , the router act like a DCP (ISDN- Device Control Protocol) server and listen (DCP messages) in 2578 port.
I need o know the contents of the TCP frame that carry (ISDN-TCP, the DCP messages) to put a SIP client to talk with PSTN/ISDN using the router. I want to write a software module in Asterisk that translate SIP in (ISDN-DCP) to connect the SIP Phones to the PSTN/ISDN using the BRI ports of the Cisco router. I need to now the contents of this message to dialogue with RCAPI server of the router.
If forum is the right place perhaps to put this, Could you give me a better place, a mail or other forum to receive the speciation of (ISDN-DCP)
thank you
With kinds Regards
JoÃ£o Pereira Rosa

PSTN to ISDN dialing question

We have the need for customers to connect anywhere in the world to our network, directly to our network. We currently have two seperate systems working two very different missions, but a similar operation. They both have to provide secure dial-up access, one is a mobile satellite system, and the other is setup for secure RAS for state side dialing.
How does it work when I dial from a PSTN line to a ISDN line, think along the lines of a direct PC to PC connection (in our case we are using secure phones to encrypt the data through a serial interface on the phone and shooting it out the phone line)
What, if any kind of problems would I run into dialing PSTN to ISDN?
Let me know if you need more info on the setup.

I forgot to mention, the analog line on the phone (it is a STE secure phone made by L-3 communications) is V.32 standard. Will that affect the connection to ISDN?

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---

Analog PSTN - ISDN

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