I have made 4 as trunk group access on the PBX for E1-PRI and 8 as trunk group access for E1-CAS.I am able to dial 4-1-6261 to place calls on telephone 6261 from 6000(outgoing and incoming over PRI). And I am able to dial 4-2-6261 to place calls on telephone 6261 from 6000(outgoing on PRI and incoming on CAS).But I am not able to dial out from my PBX extensions over the E1-CAS card using 8-1-xxxx or 8-2-xxxx. I have patched two PBXs back to back on their E1-CAS ports and am able to dial out using 8-xxxx. So this means the trunk group allocation on PBX is working fine. The show controller e1 0/0/0 is showing normal stats.When I do a show voice port 0/0/0:1 I see that one of the ds0 timeslots are being seized when I try to dial out on 8-x-xxxx but the Out Status column entry against the timeslot says clear_bak.When I am placing calls on PRI, I don't see any such indication against the timeslot being seized.Basically since I am able to dial in to my E1-CAS port, the line coding, framing, signaling(to some extent) must be right, or so I guess. But am not able to dial out.
Have a asterix PBX running my system and I upgraded my security with a cisco ASA 5505. Now all the extensions are working including the remote once. Everything elase like internet.Other servers all working fine. Only problem is that when ever someone dials a landline number from an extension it does not go through.seems like the firewall is blocking it but I cannot figure out why or how. All the NAT and Access list is fine. Although I have no idea how to accept the SIP PROXY IP through the firewall and I am guessing that might be the problem. There is no any other problem and I am 100% satisfied with the ASA5505 except this problem
We recently upgraded our ASA to 8.3, most everything went ok, but I am having problems with outgoing nat. It seems that when one our systems that needs to be natted to an outside IP address when connecting out is not doing it. When that system goes out the ip address is our internet IP and not the natted address, however, inbound everything works.
We have one rule that does PAT
nat (INSIDE,OUTSIDE) source dynamic OG_IP_NAT_DMZ obj-22.214.171.124This is the natting statement that should be translating the addressesobject network obj-10.200.0.10 nat (INSIDE,OUTSIDE) static 126.96.36.199I think I need to double nat, is that right if so how?
We have 2 h323 IP phones (non cisco) that connect to an IP PBX via a VPN connection using PIX 5505's at each end. The VPN seems to work fine and the phones can be used normally with good call quality, display, status...The problem is that when talking on the phones for any extended period of time, the phone reboots, drops the call and then restarts automatically. This problem usually happens on phone calls lasting 20 minutes or more but has happened on calls as short as 15 minutes. I have also made calls which have gone 45 minutes before the reboot. I have removed everything from the network hosting the remote IP phones except for a single phone and the problem still occurs. It is not a POE Issue as the phones can use a power brick and it still happens. The same phones work perfectly on-site when not using the VPN.Someone has mentioned that I should adjust the UDP Timout Timers of the VPN. I don't see any UDP timers and am at a complete loss on what is causing this. The data traffic does not seem to get dropped when connected accross the VPN for hours at a time.
I have router which has two physical interfaces Gi0/0 and Gi0/1. G0/0 connects to metro over ethernet and Gi0/1 is configured a s router on a stick, which has many defined. All those interfaces have IP addresses assigned. EIGRP is configured between other metro sites. Here is a sample IP assigment for this site, let's say Site.
I am encountering delays when making any calls using VOIP. I understand that there is always a delay using VOIP but not as much. I've done a bandwidth test on my network and everything seems to be Ok, we have 4 T1s with 55 Reps using it. There is nothing choking up the network.
within ACS 5.3, I'd like to use 2 external authenticator for the same service, like vpn remote-access.For the authentication, I know I can create an identity chain, to query SecurID and then AD, in case of user not found in SecurID.For the authorization rules, I need to provider a wide vèn access for SecurID users and narrow vpn access for AD user.Are there some parameter to use in compound conditions for SecurID ?
As stated in the attached picture, my company has a remote office which its PBX was connected to the main PBX via a pair of leased lines and a pair of E1/HDSL converters. (The distance is about 2 kilometers, so the E1 was converted to HDSL and then back to E1 at the remote site)
Now, IP network is developed between two sites.We want to transfer the calls via a pair of 2811 Routers, both equipped with VWIC2-1MFT-T1/E1, as the second part of the picture. I tried to find an straightforward document, but I was unsuccessful.What configurations should be made at both ends to transfer the calls to/from the remote site ?
cisco 2651XM router with WIC1 adsl card and NM-16ESW switch IOS: c2600-ipbasek9-mz.124-23.bin
I use the following config to export traffic from the adsl card to a fasterthernet port so I can look at the adsl traffic in wireshark on a pc:router(config)#ip traffic-export profile my_rite router(conf-rite)#int FastEthernet 0/0 router(conf-rite)#bidirectional router(conf-rite)#mac-address abcd.efgh.ijkl (mac address of PC) router(conf-rite)#exit router(config)#int dialer0 router(config-if)#ip traffic-export apply my_rite this config works and I can see stuff going on in wireshark but it's only one way. This config only shows traffic going out from my adsl card, but no incoming. There is defintely traffic going both ways because everything about my adsl connection is working perfectly. I've tried using a different fastethernet port, even tried exporting to a different pc but all I see is outgoing ie: source is my public ip address but never as destination . I have bidirectional in the config but it still only shows outgoing. I even tried a different IOS (c2600-adventerprisek9-mz.124-15.T8.bin) but still it doesn't show incoming traffic. Could it be my ISP in some way hiding incoming traffic from view?
is it possible to block outgonig multicast L2 frames on an Ethernet port in outgoing direction on a 2960 Switch?
I tried the "switchport block multicast" command, but the description of this feature relates to only "unknown" multicast!?
But what means "unknown multicast"? Even if activated, I see a lot of multicast traffic going out that port: IGMP, PIM, SSDP, HSRP, OSPF, .. and also pings and VLC streams to multicastaddresses (ip igmp snooping disabled).
I also tried to map a "mac access-list" to that port, but the "mac access-group" interface command is restricted to only incoming traffic.
Reason: we assume, that there are a couple of specific enddevices, that might react strange to some multicast. Therefor we would like to block outgoing multicast on that specific ports.
we've buyed a WRVS4400N to create a IPSEC VPN tunnel to our client in order to access some applications.
After a while trying to configure the router, we have archieved it and the VPN tunnel is up. We can see the tunnel up from here and from client's side as well. Our client supposendly have created the tunnel in order to access a list of specific IPs in the range 10.113.x.x, but if we try to access this IPs via telnet whe cannot obtain any response.
Making a tracert, we obtain... C:UsersHuexxx>tracert 10.113.56.177 Traza a 10.113.56.177 sobre caminos de 30 saltos como máximo. 1 1 ms 1 ms 1 ms 192.168.0.1 2 * * * Tiempo de espera agotado para esta solicitud. 3 * * ^C
... and therefor the client doesn't receive any packet at its firewall.
I've tried to establish a static route for 10.0.0.0 255.0.0.0 to their remote gateway, but I'm unable to add any entry to static routing list... The router tries to do something, but afterall I cannot see the new entry...
What can I do to route the traffic through the tunnel?
We have Cisco 1900 Series Integrated Services Routers (has a wired router and a wireless router) and since this morning we can not send emails.I inquired with both the ISP and the hosting provider and all settings are correct.I can receive emails from outside the router, but can't send any emails out.If I try to telnet mail902.opentransfer.com 25 it doesn't connect.Port 25 is the port we were using all the time and was working through the router.I connected my laptop directly to the modem and was able to send emails using port 25, also was able to telnet to outgoing mail server. I didn't change anything in the router.is there a way to "enable" port 25 or "enable" mail.homeserviceclub.com (smtp server) or mail902.opentransfer.com (hosting mail server) if this would solve the problem?I don't understand why is this happening as I never had to enable or disable any email ports or mail server addresses.
I have a CISCO 1841 ROUTER and sins short our internet speed has decreased dramatically , it does not happens all the time , so I am sure it is not the ROUTER.I have put a small router (CISCO WRT 610N) and it was the same.When I look to the UP and DOWNLOAD GRAPH from my ISP , I see really BIG peeks.
When i try to active the Internet Access Police with Website Blocking by Keyword, the router WRVS4400N block any access to internet, the Access Restriction by time is disable. How i can active this feature without restrict all the access?
We have setup the IP phone proxy on our ASA-5520, we had a couple of issues with the initial setup, but nothing major. It has been up and running for a few weeks and basically everything works perfectly just like we designed it except for 1 strange audio issue on outbound calls. We can make a call to anywhere, no problem, if the call is answered, no problem, perfect call setup and good quality 2 way audio. But if the person we called doesn't answer the call and that call goes to their voicemail we loose all audio from that point forward, we do not hear their outgoing message or get any prompts just dead air. The same situation appears to be true for any "recorded" service on the other end of the call.
I'm a new user of VOIP. My network connect is Cable modem -> router (Dir 615) -> ATA (SPA122) . Outbound calls have no problem. But inbound calls sometimes work, sometimes not ( got message: not in service)
I have the latest Firmware version 2.05NA on the DIR-825 and while I can make outbound SIP calls I can't receive any inbound calls. The phone will ring but I can't pick-up the call. [code] I am at a loss as to how to register my phone. When I uncheck ALG SIP the phone will not register to my server.I have a Polycom 550 connected to a Trixbox server. The prior router worked great.
Something a little odd happened the other night. I had spent the afternoon updating all necessary programs where updates were available, did a GRC shieldsup test, did a clean and test of my system with AVG and MBAM. Everything looked fine.I went online that night however and as soon as I went to my online banking website, I noticed that there was an outgoing attempt logged in my firewall (ZoneAlarm). It was blocked. The IP address is 188.8.131.52:80 - which apparently fits in GoDaddy's IP range. Googled it and saw that someone else had the same issue.
Trying to split a supplied fixed IP address to multiple wireless devices so that I can piggy back on the internet connection in my office. Cause the IT dept refuse to provide a router.I plank to use a router for the job above.
how I could possibly get all my e-mails of which I send my clients through my outlook to be registered on my Home-outlook, Office-Outlook and my mobile-outlook? like on the server? something similar to when you open yahoo or G-mail from any given location you can view your sent Items,inbox etc.
I heard from customers calling into the office stating that they have experienced some issues with our prompts and the extension input, that it is not recognizing their input. Stating sometimes it does not work on their land line but once they call back on their cell phone it works. We in the office all had people from the outside call from a land line to our office to verify the issue but the calls went through. We are thinking that they may be dialing the wrong extensions.Has anyone had any issues with land lines having trouble calling in.I am running cucme connected to a sip trunk.
I have CUCM 8.5 in a lab environment that connects to a Cisco 2801 which has a VIC2 FXO card on board, I have believe that I have configured the MGCP correctly and also configured my CUCM correctly but I'm just getting a fast busy when I make a call through the PSTN, I have tried removing the Ethernet cable from the Router just to test what happens when I dial external and again I get fast bust.
Here is my configuration setting: -
From CUCM gateway details: Gateway Name MGCP Protocol MGCP AALN/S0/SU1/0@MGCP AALN/S0/SU1/1@MGCP AALN/S0/SU1/2@MGCP AALN/S0/SU1/3@MGCP
I have an SA520 that is being used as a front end firewall. Behind it I have an IP PBX. The VOIP provides are registered and I can make outgoing calls. However It appears that the SA520 is either blocking or not routing the calls. I have opened the ports recommended by both the IP PBX and the VOIP provider. What do I need to do to make incoming calls through the SA520?
We're having some numbers ported over to us and we'd like to verify that we are in fact receiving all of the numbers at our SIP gateway. Since we have been getting more and more activity on this router, I'm becoming more concerned about using certain debug's for fear that the router maxes out CPU and drops.
What the best debug command would give me this information with minimal impact on CPU? In the past I've used 'debug ccapi inout' and a couple of others similar to that. With so much activity though sometimes it bombs the router. Also I am logging to the console directly, maybe there is a better method with less CPU impact. I just don't want to have to go back and forth to look for stuff 50 times either if I write out to a file or something, it could work though I suppose.
I'm in charge of the IT of a dorm with 200 inhabitants. A lot of people complain about the bad quality of "skype-calls". Audio is delayed and video stucks. The actual speed of our internet connection should be fast enough so I guess it has something to do with the QOS setting. I checked it on our WLC 3750 and right now we use Silver (best effort). Could there be any improvement when I set it to Platinum?
We have a Cisco Router 3825 which we use to convert phone calls from PSTN to SIP.Can any one suggest us (or point to a documentation) as to how can we use this router as a load balancer (to balance load among different SIP terminals) ?
I know the problem lies in the timeout for the default inspection rules. I cannot find where to change this.I have nat statements for 1 to 1 nat to my polycom and tandberg video devices. I have set the firewall for the ouside to in rules to inspect all. This allows the devices to communicate with public addresses, but the video calls drop after 1 hour. How do I change the settings so this does not happen. I have tried setting the rule to allow instead of inspect, but when I do this, I have no access at all. I just want to nat 1 to 1 and allow all traffic to the nated devices. I know this is not secure but it is the only way to allow h323 through to let the video conferencing devices work.