Endpoint VC PSTN call

Similar Questions

• Transfer to PSTN calls on the SPA3102 VOIP does not work with voip providers who do not proxy outbond

  I use the SPA3102 to transmit calls from PSTN to Voip network. Work very well with a local Voip provider calls Vono (www.vono.net.br), I think that this provider is the one who has the outgoing proxy. However, I would like to use the SPA3102 forwardin PSTN calls with other providers such as Freevoipdeal, VoipWise, VoipRaider etc... These suppliers have no outbound proxy. How can I get around this? According to the link below, you need to plan an outbound proxy. What should do?

  http://www.Cisco.com/en/us/products/ps10024/products_qanda_item09186a0080a359ae.shtml#q

  For example, freevoipdeal voip provider guides that leave the field blank outgoing proxy as below.

  https://www.freevoipdeal.com/SIP

  You can leave the outgoing Proxy field blank.  You do not have to have an entry in this field for the SPA3012 works correctly.

• Endpoint collaboration - Protocol call by default Auto - does not seek after failing with SIP H323

  Hi all

  Hoping someone in the community may be able to explain how the "auto" setting for the default calling Protocol?

  My understanding is that, when together, he should try to SIP and H323 - however, it seems to just try SIP then stop.

  I've seen CSCum27379, but could not know if it was fixed in the end.

  Context:

  We are currently move to SIP from H323, and have all items on the same VCS cluster endpoint (Polycom HDX & Cisco) double (first H323 / by default). The problem is that some people dial still using IP addresses, however our new external gateway and bridge services cannot be connected to the use of SIP (non-interoperable).

  The Polycom HDXs are ok, they'll just try a protocol after the other.

  The Cisco seems to be stuck on one, so either not will have access to the new services (except preconfigured in the address book), or will have all their IP address interoperability of point to point calls.

  Is there a way to direct address IP SIP work while composition on the VCS?

  I was also hoping to create a search rule that blackholed IP SIP addressed calls would force the reversal, but did not: (.)

  As always, appreciate any guidance you can provide.

  Rich

  This bug link you to, CSCum27379, there is no solution for it that its status is "completed".

  There is a specific order in which the endpoint will attempt to establish a call and depends on what protocol endpoint is saved for use, pg 84 SX20 Administrator's Guide :

  Allows the automatic selection of the memorandum of Appeal based on what protocols are available. If several protocols are available, is of the order of priority: 1) SIP; (2) H323; (3) H320. If the system is unable to register, self-selection chooses H323

  Endpoint won't try the first Protocol in the list and then switch to the other, he tries only one he can and that's all.  If you subscribed using SIP, but you need to make a call to a registered endpoint H323, you must set the Protocol to call to serve as a H323 or allow interoperability on the VCS to allow SIP-H323 calls.

• DX 80 - dial PSTN call does not

  We have setup in the video below. But in recorded in CUCM 10.5 x SX80 try calling PSTN numbers, does not.

  Hit gateways voice with error, error of unavailable resource. But other devices work properly with the same number of rank. Help, please

  Hello

  Can you put the codec g729r8 on your existing voice class codec and affect your pointing to your (not only G711ulaw) CUCM voip dial-Exchange.

  Also configure transcoding on router resources and register with CUCM.

  In my view, that it is somewhat related to DSP.

  Kind regards

  Raclot.

• Incoming PSTN call to fail UCCX

  Hello

  I have no call flows as below

  RTC - sip - CUBE (SP edition) - sip - CUCM - CTI Port - CCX

  When RTC called the number and it should trigger CCX application to read the message BUT rather we transport message ' xxx notifies you that the number tried to call has been disconnected. "

  Track of the CCM shows 404 not found but DNA shows the works of appeal show. Why?

  CCM traces and traces of CCX engine is attached.

  CM version is 10.5

  Version of the CCX is 10.5

  call: + 61423348232

  call: + 61730842590 (DN 41490 is CTI RP CCM registry).

  DN 40700 is CTI port.

  CTI port 40700 and CTI RP 41490 are attributed to jtapi user application in CUCM

  FH

  Make sure that the SIP trunk CSS lists partition associated with the ports above the Point of road ITCS ITCS even if the call goes to the CTI RP you need access to CTI ports for the call to connect.

• SPA-3102: how to fill the calls from SIP to PSTN and vice versa?

  Hello, since my ISP to my office blocked SIP ports, of I'll try and use it at home where the ATA works very well.

  If I call you ATA installed in my office in the United States (where there is no problem) to my ATA at home in India, I have how to route the call selectively to theFXS port or port FXO of the anti-terrorism Act in India? I I want to answer the call directly using the phone connected to the ATA instrument or make another local call out on the PSTN connection.

  TIA

  You get two accounts of Gizmo.  You put an account on the tab line 1, you put the other account on the PSTN line tab.  If you want to ring the phone attached to the SPA3102, you dial the account tab line 1.  If you want to get the tone to fill an outgoing line PSTN call, you call the account registered on the PSTN line tab.

• Branch call PSTN routing

  Client of HQ and Branch office with 512 line MPLS to only HQ they had E1 trunk, branch to call PSTN the call had to go via MPLS in HQ and the call RTC branch terminated at HQ only, is it a good design or better we need to add an ISDN line at the level of the branch to end the PSTN call.

  in terms of good design, it is not

  because this design has some imitations like below

  1-single point of failure, if the MPLS link is near the bottom of the branch cannot be accessed via PSTN and they will lose access to the PSTN, too

  2 - the MPLS link will be as a bottleneck for this branch because the PSTN and onnet calls will go on the PSTN connection

  However if this is due to the customers needs/budget then you can do it like that, but don't forget to mention the recommendations for local PSTN

  HTH

  If useful rates

• Disconnection of call for the endpoint MXP

  Hello world

  We have 1 special issue of one of our series mxp endpoint, where the call disconnects in between it the call, we call for our local endpoint in India on another mxp parameter to the United Kingdom on a dedicated IP link.

  The two endpoints are registetered to a common gatekepper.

  IM attaching the Syslog and Syslog 4 of our local endpoint 3 mxp mxp for your reference, please check the logs & suggest what could be the reason of disconnection of the call here.

  Thanks in advance

  Nikhil

  Hi Nikhil,

  Newspapers showing your EP get no response from the return of unity to end a call is disconnected.

  This problem is the network problem, check firewall, router configuration.

  H.245 Peer H323CC-1 entity does not meet the demands of round-trip time

  H.245 Peer H323CC-1 entity is presumed down. Disconnectiong.

  Kind regards

  Dharmmesh

• Does anyone know why an edge 95 MXP will call himself?

  I have an Edge 95 MXP and call himself all the time... the call is a SIP call of [email protected] / * /a call of 64K and the IP address. I don't have the power off because it's in the ceiling.  The unit is also recorded as a 3rd party endpoint call Manager 10.5... The unit can make calls at our other fine video devices. I had to set the unit to auto-answer, so it does not sound at all times.

  Usually, it is because your getting analyzed by applications such as SIPVicous tent identify opportunities for exploitation of circuits on UDP SIP PSTN.

  In general, this risk is mitigated by closing the firewall ports for the endpoint or by making a few changes to its configuration.  On a codec of series C, EX, SX, MX running TC software that is easy, but I do not know the equivalent commands (if there is) for a codec MXP.  However, you should be able to block the scans that cause this by closing the port UDP 5060 if the codec is open on your firewall.  If she goes by a gatekeeper as a VCS server you could disable UDP, I do not know CUCM.

  This has been asked several times, do a search in the forums for "100 @" found the following results, although they all have to do with a series C or VCS codec, but it's all the same question.  The unit is publicly accessible via the internet, it's just a matter of closing or adjustment of the endpoint configuration or call control server.

  lot-Calls-Numbers-100-and-101-looks-Self-Calls

  strange-appeal-loop-c40

  Cisco-VCS-Alarm-Call-License-Limit-Reached

  C20-phantom-appellants

• Crossing call - 403 forbidden problem

  Hello

  We deployed the VCS control with VCS Expressway for the course of firewall and encountered a problem. It says on the status of the call VCSe, 403 forbidden

  When an external end point registered as SIP on the VCSe will call to an internal endpoint as a SIP to the VCSC.

  Here are some of the scenarios test, we have carried out and the results:

  1. check internal endpoints SIP VCSC - OK

  2 external endpoint SIP registartion to VCSe - OK

  3 internal endpoint endpoint internal SIP calls via VCSC - OK

  4. external point endpoint endpoint external SIP calls via VCSe - OK

  5 internal endpoint endpoint external SIP calls through the VCSC and VCSe - OK

  6. point endpoint endpoint internal external call SIP via VCSC and VCSe - failed (403 forbidden)

  On item 6, which is the only problem we are trying to solve.

  Which are necessary to check?

  Thank you.

  BR,

  Acevirgil Ocampo

  I recommend no doubt create a subzone for Movi, you can also control the bandwidth also. I have never save anything to the default subfield on Highway and off the record for security purposes. Movi clients enter their own subfield

  NES as well as sub-areas for other specific things.

  For security reasons, I don't like is not verification of the credentials on the highway, especially if you have an ISDN gateway or maybe control has a route on the PSTN. Fraud free of charge is not sweet.

  I always sent SSO via local vs LDAP authentication. It becomes difficult when you are unable to join the fast track to the area but still want to authenticate and verify the credentials. The trick is that you must create a local user on the highway, and then create the same identical user name and the password in active directory. Then in MSD, get these credentials by using the model configuration for the version / MOVI device.

• SPA3000 dial of Voip to PSTN problem

  
  

  Do it with the dial plan.  Set the PSTN Caller default DP a number available and configure the dial plan:

  (S0<:[email protected]>) where [email protected] is the sip uri when you want to transfer the call.

  If it's a PSTN number you want to transfer the call to you must configure tab PSTN line with the credentials of voip to voip account, you want to use to transfer the call.

  Of course, the line 1 tab you set enable component IP: Yes, if you send to a sip uri.

• SPA3102 problem: State PSTN: still rings in the State

  We got a new router Linksys SPA3102 VOIP gateway, and we have some problems to operate.

  The SPA3102 is properly connected to the local network, but it seems to have problems with the incoming PSTN line.

  However, it does not sound when I call the number RTC. On the other hand, State PSTN always says 'Ringing '...

  When I restart the SPA3102, then the extension on the pbx sipx she must call rings (without anyone calling the PSTN number).

  The status page says something like:

  Voltage: 14 (V) loop current: 0.0 registration of (my) State: not recorded the last record to: next record in: last call VoIP number: [email protected] last call number RTC: last calling VoIP: last PSTN Caller:, last RTPC Disconnect reason: CPC Signal RTC activity Timer: 30000 (ms) mapped Port SIP: 5061 Type of call: VoIP Gateway call VoIP State: Idle PSTN State : ringer VoIP: tone RTC:

  (Note: the bridge didn't need to register for the sipx-pbx).

  While it fires after a reboot of the SPA3102, nothing happens when I dial my mobile PSTN - I hear only ringing on mobile (it says busy or more).

  Does anyone have advice?

  Andreas

  It did not break. The problem was with the cable to connect to the UK British Telecom. I haven't used the gray wire that came with it, but used a telephone cord UK I had to hang out - bad move.

  When we talk about Linuxemporium, there where I bought the SPA3102, I was made aware that the cable might be the problem. So I went on the road in a small phone shop to get a US - UK adapter (packaged with the SPA3102 is still an American cable), that didn't fix it. I had another well-known adapter from a retailer of Electronics UK, and that one has solved the problem. The SPA3102 fortunately now works as my PSTN gateway.

  I hope this can help others with similar problems.

• SPA3102 - disable incoming PSTN

  I have an SPA3102 I am putting in place so that

  -ATA outgoing calls through voip (work)

  -incoming calls from the PSTN are ignored

  -outgoing emergency calls 911, etc. are out by RTC (work)

  -outgoing calls if VOIP connection downwards through the PSTN. (work)

  I can't seem to find a way to have it ignore incoming PSTN calls.  I have this configuration in a suite of grandmother, as she makes calls sparingly, but we wanted to the PSTN where the voip connection is broken.  Now that I have everything set up, every time there is an incoming call on RTC, it sounds the phone on the ata...

  Is there a way to disable this?

  Thank you

  John

  JRoesner wrote:

  -incoming calls from the PSTN are ignored

  I can't seem to find a way to have it ignore incoming PSTN calls.

  Is there a way to disable this?

  On the tab game ring Thru line 1 PSTN PSTN line: No.

• SQL query for extension able to make NET calls

  Hello

  Is it possible to create a query that can retrieve only the extensions that are able to PSTN calls?

  I am currently interviewing peripheral and numplan for all devices whose device.name is like MS %. However it will only retrieve the IP phones.

  I need to create a query that can recover all other types regardless of what looks like their name of the unit. I intend to run it on 10.5 CUCM and 11.0

  Thank you

  This should help you get the result required that you are looking for.

  http://www.ucguerrilla.com/2012/04/CUCM-SQL-queries-listing-call-forward...

  BR,

  Nadeem

• duplicate calls made using the alias between two codecs C40

  We have 2 new codecs C40.  When calling between the two using the alias, it seems that 2 calls are made.  In a sense, we get a very quick disonnect message upon acceptance of the call.  In the other direction, the call comes in, but on disconnect, we get a message saying missed call.

  Seems that the search rules can be duplication of the call some how - probably him endpoint becomes a call SIP and H.323 call at the same time. Could you post the call details and history/details of research?

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