Call RTC SRP541w

Anyone know how I can make a call via it to PSTN port when the phone is connected to a voip port. I guess I have to put some kind of code in the phone which will not fail on the PSTN port router so that I can make the call via PSTN and not through my voip provider.

Hi Jim,.

Please see the following post, I hope that answers your question.

https://supportforums.Cisco.com/message/3755255#3755255

Kind regards

Andy

Tags: Cisco Support

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    -[358291@from-internal:1] Macro execution ("SIP/205-087af3d8","user-callerid |") SKIPTTL | ") in new stack
    -Execution [s@macro-user-callerid:1] Set ("SIP/205-087af3d8", "AMPUSER = 205") in new stack
    -Execution [s@macro-user-callerid:2] GotoIf ("SIP/205-087af3d8","0?") report") in new stack
    -[S@macro-user-callerid:3] ExecIf execution ("SIP/205-087af3d8","1" | ") The value | REALCALLERIDNUM = 205 ") in new stack
    -Execution [s@macro-user-callerid:4] Set ("SIP/205-087af3d8", "AMPUSER = 205") in new stack
    -Execution [s@macro-user-callerid:5] Set ("SIP/205-087af3d8", "AMPUSERCIDNAME = Damir here") in new stack
    -Execution [s@macro-user-callerid:6] GotoIf ("SIP/205-087af3d8","0?") report") in new stack
    -Execution [s@macro-user-callerid:7] Set ("SIP/205-087af3d8", "AMPUSERCID = 205") in new stack
    -Execution [s@macro-user-callerid:8] Set ("SIP/205-087af3d8", "CALLERID (all) ="Damir Reic ' <205>' ") in new stack
    -Execution [s@macro-user-callerid:9] Set ("SIP/205-087af3d8", "REALCALLERIDNUM = 205") in new stack
    -[S@macro-user-callerid:10] ExecIf execution ("SIP/205-087af3d8","0 |") The value | Channel (Language) = ") in new stack"
    -Execution [s@macro-user-callerid:11] GotoIf ("SIP/205-087af3d8","1?") continue ") in new stack"
    -Goto (macro-utilisateur-callerid, s, 20)
    -Execution [s@macro-user-callerid:20] NoOp ("SIP/205-087af3d8","Using CallerID"Damir Reic" <205>" "") in new stack
    -Execution [358291@from-internal:2] Set ("SIP/205-087af3d8", "_NODEST =") in new stack
    -[358291@from-internal:3] Macro execution ("SIP/205-087af3d8","record-enable |") 205. OFF | ") in new stack
    -Execution [s@macro-record-enable:1] GotoIf ("SIP/205-087af3d8","1?") check ") in new stack"
    -Goto (macro-record-enable, s, 4)
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    -Launch Script AGI/var/lib/asterisk/acted-bin/recordingcheck
    recordingcheck |-205946 20090827 | 1251399586.64: outgoing recording not enabled
    -Recordingcheck AGI completed Script, return 0
    -Execution [s@macro-record-enable:5] MacroExit ("SIP/205-087af3d8","" "") in new stack
    -[358291@from-internal:4] Macro execution ("SIP/205-087af3d8","dialout-trunk |") 2. 358291 | ") in new stack"
    -Execution [s@macro-dialout-trunk:1] Set ("SIP/205-087af3d8", "DIAL_TRUNK = 2") in new stack
    -Execution [s@macro-dialout-trunk:2] GosubIf ("SIP/205-087af3d8","0?") Sub-pincheck | s | 1 ") in new stack"
    -Execution [s@macro-dialout-trunk:3] GotoIf ("SIP/205-087af3d8","0?") disabletrunk | 1 ") in new stack"
    -Execution [s@macro-dialout-trunk:4] Set ("SIP/205-087af3d8", "DIAL_NUMBER = 358291") in new stack
    -Execution [s@macro-dialout-trunk:5] Set ("SIP/205-087af3d8", "DIAL_TRUNK_OPTIONS = tr") in new stack
    -Execution [s@macro-dialout-trunk:6] Set ("SIP/205-087af3d8", "OUTBOUND_GROUP = OUT_2") in new stack
    -Execution [s@macro-dialout-trunk:7] GotoIf ("SIP/205-087af3d8","0?") nomax ") in new stack"
    -Execution [s@macro-dialout-trunk:8] GotoIf ("SIP/205-087af3d8","0?") chanfull ") in new stack"
    -Execution [s@macro-dialout-trunk:9] GotoIf ("SIP/205-087af3d8","0?") skipoutcid ") in new stack"
    -Execution [s@macro-dialout-trunk:10] Set ("SIP/205-087af3d8", "DIAL_TRUNK_OPTIONS =") in new stack
    -[S@macro-dialout-trunk:11] Macro execution ("SIP/205-087af3d8","outbound callerid |") 2 ") in new stack"
    -[S@macro-outbound-callerid:1] ExecIf execution ("SIP/205-087af3d8","0 |") SetCallerPres | ") in new stack
    -[S@macro-outbound-callerid:2] ExecIf execution ("SIP/205-087af3d8","0 |") The value | REALCALLERIDNUM = 205 ") in new stack
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    -Execution [s@macro-outbound-callerid:7] Set ("SIP/205-087af3d8", "EMERGENCYCID =") in new stack
    -Execution [s@macro-outbound-callerid:8] Set ("SIP/205-087af3d8", "TRUNKOUTCID = 383597") in new stack
    -Execution [s@macro-outbound-callerid:9] GotoIf ("SIP/205-087af3d8","1?") trunkcid ") in new stack"
    -Goto (macro-sortant-callerid, s, 12)
    -[S@macro-outbound-callerid:12] ExecIf execution ("SIP/205-087af3d8","1" | ") The value | CALLERID (All) = 383597 ") in new stack
    -Execution [s@macro-outbound-callerid:13] GotoIf ("SIP/205-087af3d8","1?") output ") in new stack"
    -Goto (macro-sortant-callerid, s, 11)
    -Execution [s@macro-outbound-callerid:11] MacroExit ("SIP/205-087af3d8","" "") in new stack
    -[S@macro-dialout-trunk:12] ExecIf execution ("SIP/205-087af3d8","1" | ") AGI | fixlocalprefix") in new stack
    -Launch Script AGI/var/lib/asterisk/acted-bin/fixlocalprefix
    > fixlocalprefix: using chart 0 + [0] [9] X.
    > fixlocalprefix: using mires NXXXXX 0 +.
    == fixlocalprefix: Dialpattern 0 + NXXXXX matched. 358291-> 0358291
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    -Execution [s@macro-dialout-trunk:13] Set ("SIP/205-087af3d8", "OUTNUM = 0358291") in new stack
    -Execution [s@macro-dialout-trunk:14] Set ("SIP/205-087af3d8", "custom = SIP/pstn") in new stack
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    -Execution [s@macro-dialout-trunk:19] Dial ("SIP/205-087af3d8","SIP/pstn/0358291 |") 300. ") in new stack"
    -Called pstn/0358291
    -SIP/pstn - 0885 has 138 sounds
    -SIP/pstn - 0885 has 138 replied SIP/205-087af3d8
    -Packet2Packet bypass 087af3d8/205-SIP and SIP/pstn - 0885 has 138
    -Remote UNIX connection
    -Connection to UNIX distance disconnected
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    -Execution [s@macro-hangupcall:3] GotoIf ("SIP/205-087af3d8","1?") skiprg ") in new stack"
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    -Execution [s@macro-hangupcall:9] GotoIf ("SIP/205-087af3d8","1?") theEnd ") in new stack"
    -Goto (macro-hangupcall, s, 11)
    -Execution [s@macro-hangupcall:11] Hangup ("SIP/205-087af3d8","" "") in new stack
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on "SIP/205-087af3d8" in the macro 'hangupcall '.
    == Spawn extension h (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/205-087af3d8.
    == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/205-087af3d8"macro"dialout-trunk.
    == Spawn extension (of-internal, 358291, 4) exited non-zero on 'SIP/205-087af3d8.

    As you can see he composed, if found the dialing rule, but nothing happens on the SPA, it will not simply compose anything on.

    I guess that something is wrong in RTC spa or configuration of trunk, but I don't know what. I tried like 5-6 guides of strach and none worked for me. Any suggestions?

    Kind regards

    Damir

    HUH! I won't curse now, but I shoul!

    Given that I'm NOT using any VOIP provider and I JUST wanted to use RTC it makes SENSE I don't connect anything to the WAN port, BUT when I configured the WAN port with my LAN IP (172.16.1.200) and put a few other IP on the LAN port, everything started working.

    It's rly rly rly is NOT logical, but I got it working now.

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    The other way, however, does not work: I get syslog entries that the call is detected, but the device doesn't send what anyone on the server (as checked with wireshark)
    The dial plan is
    (S0<:1234>)
    but I also tried
    (S0<>[email protected] / * />)
    and some variations more

    This is syslog entries:

    FXO:start CNDD
    Number of the caller analysis = callingnumber
    -Caller ID:
    -Name = (null)
    -Number distance = callingnumber
    -Dialable number = (null)
    -No reason number = (null)
    FXO:CNDD = name, number is callingnumber
    FXO:stop CNDD
    Phone = FXO:CNDD name = callingnumber
    Your RTC AUD:Stop
    FXO: on the hook
    Your RTC AUD:Stop
    the next sequence is repeated several times, probably as long as the line is actually ringing
    FXO:start CNDD
    Your RTC AUD:Stop
    FXO: on the hook
    Your RTC AUD:Stop
    FXO:stop CNDD

    Where would I look next?

    I found a tip in an ongoing discussion, and in fact this has solved the problem: "PSTN ring timeout" must be longer than the time to ring + break from the ring, and "Time of response to PSTN" must be short enough

    Unfortunately, if incomplete description of configuration. There are so many dial plans to set, but it has not specified that have configured it.

    Please read before RTC call the SPA3102 to VOIP. I hope this will help you.

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    We have a Manager call we use for AA to provide wishes and staff of mailbox access mailbox option and no DDI extension before you send us the receptionist (Call Manager driver Point to the monitoring console). Greeting for office hours and the other for off hours which takes messages.

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    TIA

    D

    Check out this thread for more information on how to replace the system with a white prompt we so used to hear you it any longer:

    http://forums.Cisco.com/eForum/servlet/NetProf?page=NetProf&type=bookmarks&CommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40%5E1%40.ee8253b/1#selected_message

    do not forget that it is not taken in charge of TAC, and the change will be deleted when you upgrade so you don't have to remember to reapply after.

    Not that you can do on the Department of HEALTH it - it is paid by the switch, not the unit.

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