Cisco :: Cucme Through Sip Trunk Not Receiving Calls From Landlines
Apr 9, 2013
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've looked in many places but cannot see how or if it is possible to configure a phone, in CUCM to have a feature ring instead of the normal ring.In CUCME you go into the ephone x configuration mode, and assign the DN to the phone with the button xfx command. What this gives you is a slightly different ring tone when a call comes through. If I am not mistaken it is the same ringtone they use on the show "24".Is there a way to do this "feature" with CUCM?
Is there really any reason why you wouldn't use spanning-tree portfast on a trunk port other than a trunk between two switches? We have it enabled on all ports except for the fiber trunk between two non-stacked switches and the trunk ports connected to our Astaro firewall.I'd like to enable it on the ports to the firewall unless that would cause issues.
we recently aquired a managed services job and have to do a overhaul of the vlan configs and have a whole dozen WC2948G's trunk between a set of ports as well as trunk out a LAG channel setup to non cisco equipment. the deal is the lacp-channel works properly on both ends but no routing of vlans between ports and between the lag trunk are working.
theres alot of settings in the config and im planning on clearing it and starting from scratch but before i do i want to know where my problem lies.
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 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.
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 ?
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.
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.
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
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
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.
About 2 years ago I bought my first laptop and the person I was with recommended my purchasing of a Belkin router. I bought the router, it worked great until this past summer when suddenly whenever somebody called my house phone, my connection was dropped. I was angry cause naturally I have those family members that want to call and talk on the phone with my mother for hours. Eventually I figured out I could work about the problem by hooking the Ethernet cord on my Westel directly to my laptop. Its a temporary fix though in my opinion because this requires me having to sit on the floor against my bed room door and this isn't comfortable, not to mention its also confining, I miss walking around the house with my laptop. Anyway this past weekend I decided to try buying a new Belking N300. This however didn't solve the problem, I still got my signal dropped whenever somebody called the house.
We have a 100Mbps metro-ethernet link connecting to a remote site. This link is terminating in our 3750-x stack switch.We are getting a LOT of output drop on that interface and our remote IP Phone calls keep going off because of that.I've disabled all kind of QoS, but no luck.
I'm working on a project in which we have to transfer our mobile phone calls via internet and our mobile should be connected to pc via bluetooth. This is pretty easy at dialer's end but how to recieve the call via internet on some pc and direct it to the reciever via bluetooth.
I have a DSL connection and obviously it coming through the phone line. Is there any kind of software that I can use to make landline calls through my computer? The DSL modem is ZXDSL. I know I can use Skype, vBuzzer and Google Talk etc etc. But I want to make calls through my landline using my computer or laptop.
I have only seen this problem with several video conferences from one location. Currently there is only one T1 connection from our HQ to remote location. When two conference calls are set to use 256Kbps connection to come back to the main HQ, after an hour or so the T1 connection would drop. What I mean about drop is that the route on the 2911 will disappear. I have to wait about 10 minutes for the connection to come back up or I have to reboot the router for the connection to appear again. There are no interface drops on the serial connections or ethernet connections.
HQ - 2911 Remote - 1821 2911 running config: Current configuration : 3529 bytes ! version 15.0 service timestamps debug datetime msec
I have a Cisco wag120n adsl modem. I am setup with a SIP provider, so all my incoming and outgoing calls use SIP.So my question is, how does my ADSL modem handle incoming SIP calls,e.g the SIP call comes in and reaches my adsl modem. Then, how does the modem find the actual port that has my ATA adaptor (analog phone) or CME router attached ? I know port forwarding would handle this, but I am not actually using any configured port forwarding rules.
We have a new deployment where we have 5 total 7921G wifi phones connected via 2 AP541n access points, one connected to a ESW520p switch and then to the UC540 and the other connected directly to the UC540. The wifi phones are intermittently giving no audio when calling each other, however, calls to the PSTN consistently do have 2 way audio. Is this a security issue perhaps? We are running the latest CCA software pack as this is a brand new deployment, also the AP's were upgraded to the latest firmware.
I just bought an e2500 to replace my WRT54g, hoping that it would fix my call dropping problem. I had tried installing some friend recommended third party firmware on the WRT54g that he said would fix my problem, and with my infinite technological wisdom and experience ended up bricking it.
My problem is that randomly during skype calls with my family back home, I will lose internet connection. It drops the skype call and then I cannot open a web page or anything for about 60 seconds. My internets dead, then it just starts working normally again by itself. I can have anywhere from 60 seconds of connectivity up to 20 minutes or more while skyping. This all happens over a wired connection. Changing the router didn't change a thing, the problem still happens in exactly the same way. I have good bandwidth and all that stuff.Also, I watch netflix on my Xbox 360 over a wireless connection. Sometimes it will just lose connection in the middle of a show. However, the difference is that I can reconnect almost instantly after an interruption on that device. I have fixed the settings on the two different channels according to one of the other post I found in this community, and have not tried those settings out yet. This doesn't worry me as much as the wired connection issues.