Voice over IP

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Hi,I configure the as5400 with two dial peers for provider A and provider B. Both of them offer sip calls.If provider A down, it takes 5sec - 6 sec to overflow next dial peer. I check the sip-ua, there are a lot of settings. Which one can control the...

I am trying to remote into my office network from home so I can run a VOIP phone from the house on the office network. I am having trouble getting the initial settings going for this. Is this a possibility? I am fairly new to the Cisco world. Any hel...

jftowater by Level 1
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  • 1 replies
  • 0 Helpful votes

Hi,How can we know what Bc is OK when configuring rate limit in an interface? We've seen that you can divide the CIR / 4 and that could be OK but, is there any document or something explaining when to use that formula? Is the same for VoIP networks?T...

enriquebs by Level 1
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Hi Netpro,I found AA/Voicemail does not coexist with a registered sip trunk on UC500. Does anyone else have this experience?My config is as follows:===========voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-conn...

Chuan Liu by Level 1
  • 496 Views
  • 3 replies
  • 0 Helpful votes

I have an ITP running on 2651XM as an STP. I have a linkset with on link. If I want to add another link to the linkset, I will add it under:cs7 instance 0 linkset ls1 X.X.X link 0 Serial0/0:5 link 1 Serial1/0:5Do I need to change anything on PGW in o...

pax_2111 by Level 1
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Has anyone seen this before? Some channels go into Blocked state and no calls can be made in or out. The only solution is to reboot the router. Cisco is saying Telco switch is sending E_DSP_SIG_1100 signal which causes E1 channels to go into Block...

Hi,I have one AS5400XM, it handles the sip calls and routes the call to providers. The first route is provider A, second is provider B, the last is provider C. I find that the call cannot overflow to provider B and C. any missing? rgdsThe dial-peer s...

Hi Group,Can any one point me to a white paper or document for writing a QOS plan?I am in the process of building a network with VoIP, IPTV and Audio visual systems that will all ride our network, and was looking to see if any one has experience with...

rsharifi by Level 1
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I know this sounds pretty silly to most of you because you are gurus but this is beating my mind. I see companies claiming to offer VOIP minutes for termination. How do they get the minutes they offer? what equipment do i need to be able to do same?

We seem to be having RTP issues with an Adtran600 (see below) and it all seems to point to my Cat6500- Count for RTP VOICE RX is 0. Any ideas? Gateway Stats │ Gateway Link Status UpPOTS Stats │ Endpoints Active 0Endpoint Stat...

cozyk1515 by Level 1
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Resolved! Securing SIP Trunk

I have a 3845 with 2 ISDN PRA interfaces running as a PSTN gateway. Customers running various PBX systems connecting to this gateway via SIP trunks. I want certain calls from a specific customer go outof a specific PRA. I achieve this by checking 'ca...

Chuan Liu by Level 1
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