, then it will show the SIP profile “Registered”. Public username: your need to put “sip: ” it will pre append automatically) Use this to select your access point SSID which your are connected(Your router’s Wireless connection’s name). settings -> SIP settingsįrom the options select ” New sip profile” On your phone and goto Menu -> Settings->connectivity->Admin. I could successfully configure actionvoip with E6. Rule 1 /^80226/ /9725380226/-NB I have added a ^ to the beginning of 80226.Here I brifely describes how to configure nokia E6 with a voip provider. You need to modify this tranlsation rule. Remote-Party-ID: "Solomon Kavala 80226" "Solomon Kavala 80226" translation-rule 3 The call come in from CUCM with a te correct CLI 972580226 Your transation rule is modifying the CLI you are sending to VzB. So remove the voice class codec and set the codec to use g711alaw as you did yesterday There is no point using voice class codec when you have already set the codec from cucm to use g711alaw. We can see the reason for service unavailable.cause code 47 (which means codec related issue) You are doing early offer from cucm with g711alaw, and you have voice class codec configured with g711ulaw as te preferred codec.Hence creating a codec mismatch. You need to open a ticket with your service provider and find out why they cant route the number you dialled But now we see your ITSP telling us that the request timed out NB: this config will only work for the configured number.You can define a range of patterns so as to allow all outboubd calls to your sip provider match this dial-peer Configure a inbound dial-peer that matches calls to your ITSP to use sip protocol and G711a as the codec.ĭial-peer voice x voip-where X is any number you want to assign To resolve your issues you will need to adjust your config as follows:ġ. Your inbound dial-peer routing calls from CUCM is however configured for From the trace.you have a sip trunk from CUCM to voice gateway.CUCM sends an invite with early offer and G711alaw codec There are a few things wrong with your config.ġ. "'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson ![]() The second this is that your dial-peer config is still not correct.Because the call is matching dial-peer 8801 first. Translation-profile outgoing Outgoing-Intl Then apply this to the dial-peer 1002 like this.ĭescription OUTBOUND G711 Voice SIP calls to VzB Voice translation-profile Outgoing-Intl-(NB "I"=capital letter for i) If you need to strip the "0" for other calls to work, then you need to configure a new translation profile for international calls like this. This rule is applied to the dial-peer 102 and is stripping the "0". +++The reason is because of this translation rule+++++++ Voice translation-profile Outgoing-Translate Remote-Party-ID: "Solomon Kavala 80226" party=calling screen=yes privacy=offįrom: "Solomon Kavala 80226" tag=5584A984-0 INVITE sip: SIP/2.0- the leading 0 is missing from the call The call is sent incorrectly to your ITSP.
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