SPA 122 - Loopback capabilities

Hi,
I am looking at ways to test audio quality when a customer reports audio issue. The PBX can echo audio and I would like to see if the ATA could do the same so that I could originate a tone from the PBX to an ATA in loopback mode so I can listen to it and check the quality I recieve.
Ideally would be to have a command (like sending a NOTIFY to the unit) sent to the ATA to put the FXS in loopback. Or have a some physical cable we could ask our customer to plug on PHONE1 and PHONE2 so I could call one FXS and answer that call on the second line ...
Anyone has ideas about that?
Thanks,
Benoit

Yes, Linksys devices are "loopback call" capable. They can be either loopback stream source as well as loopback deflector. So you can initiate loopback call to end-user's ATA device with no user intervention. The called device will transmits the audio packets that it receives back to the transmitter/receiver instead of transmitting the data sampled on attached analog telephone. Unfortunatelly, the loopback call feature is not documented by Cisco (as far as I know). On the other side, it seems that RFC 6849 has been created with the Linksys's implementation in the mind and Cisco's employee is co-author of such RFC. So it may be relevant and may help to understand the feature details despite Linksys implementation may deviate from it a lot.
Following informations are known to me.
No, such feature is not triggered by a NOTIFY message. The loopback call is negotiated in SDP. See catched SDP bellow:
Standard call
Loopback call
(source/media mode)
m=audio 16532 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=audio 16530 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=loopback:rtp-media-loopback
a=loopback-source:8
m=audio 16530 RTP/AVP 113
a=loopback:rtp-start-loopback
a=rtpmap:113 PCMA/8000
In packet media loopback type, the loopback:rtp-pkt-loopback is used instead of loopback:rtp-media-loopback
In mirror media loopback mode, the loopback-mirror:8 is used instead of loopback-source:8 and m=...113 media descriptor is not present at all.
The id 113 is id of RTP-Start-Loopback Dynamic Payload as configured in phone setup (113 is default value). The id 8 in loopback-source/loopback-mirror is id of RTP-Start-Loopback Codec as configured in phone setup (here 8 = PCMA).
Following setup options are related to loopback call feature:
Media Loopback Code
The star code used for enabling media loopback on the phone.
The default i *03.
Accept Media Loopback Request
Controls how to handle incoming requests for loopback operation. Choices are: Never, Automatic, and Manual,
where:
never — never accepts loopback calls; reply 486 to the caller
automatic — automatically accepts the call without ringing
manual — rings the phone first, and the call must be picked up manually before loopback starts
The default is Automatic.
Media Loopback Mode
The loopback mode to assume locally when making call to request media loopback. Choices are: Source and Mirror.
Default is Source.
Note that if the ATA device answers the call, the mode is determined by the caller.
Media Loopback Type
The loopback type to use when making call to request media loopback operation. Choices are Media and Packet.
Default is Media.
Note that if the ATA device answers the call, then the loopback type is determined by the caller (the ATA device always picks the first loopback type in the offer if it contains multiple types.)
ENCAP RTP Dynamic Payload
The dynamic payload value (96 – 127) used for the encapsulating RTP packets when offering the SDP to loopback packets. This setting is used if the SPA is the offerer of the SDP. Otherwise, the value is decided by the peer.
The default value is 112.
RTP-Start-Loopback Dynamic Payload
The dynamic payload value (96 – 127) used by the mirror in the self-generated RTP packets before receiving any RTP packets from the source. This setting is used only when the SPA is acting as the loopback source. Otherwise, the value is decided by the peer.
This value must be different from any of the dynamic payload values that might be used by the source and the mirror (including the encaprtp payload type). This is necessary so that the source can easily tell when the mirror has switched from sending self-generated RTP packets to sending loopback packets.
The default value is 113.
RTP-Start-Loopback Codec
The actual codec corresponding to RTP-Start-Loopback Dynamic Payload, whose codec name is used in the rtpmap attribute for the for the mirror self-generated RTP audio stream, prior to receiving any RTP packets from the source.

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    What you're describing is indeed pretty weird. I've just tested in my lab using a factory-defaulted SPA122 running 1.3.1(003) which I registered to two different ITSPs. An inbound call to the SPA122 FXS port results in a 100 TRYING followed 0.12 seconds later by the 180 RINGING. I see no delay between the 100 and 180.
    It may be worth factory-defaulting the ATA in case its configuration has been to something unknown.
    Regards,
    Patrick---
    Use this reference document to locate SPA ATA resources

  • Spa 122 inbound problem "Number called not in service

    Hello When i call my home # for first few tries i get number is not service  then call goes thru .is there is any setting in SPa 122

    Set up the debug log so you can see if the proxy is forwarding the call to the spa122. 
    Info on setting up debug log at https://supportforums.cisco.com/docs/DOC-9862
    After setting this up, make a call and see if the call shows up in the log file.  If it doesn't, then you'll know that the proxy isn't forwarding the call to the unit.
    You can also check if the unit is/isn't registered at the time of the call.  Check the info tab of the web UI.

  • Newly fetched settings getting reset all the time (provisioning of SPA 122)

    The initial configuration works and is pretty much a sample file that sets some defaults and resets the provisioning rule to fetch an encrypted file with this information:
    <?xml version="1.0" encoding="UTF-8" standalone="yes"?>
    <flat-profile xmlns="http://www.sipura.net/xsd/SPA122" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="http://www.sipura.net/xsd/SPA122 http://www.sipura.net/xsd/SPA122/SPA122-1-3-2-014.xsd">
      <Resync_Periodic ua="na">3600</Resync_Periodic>
      <Resync_Error_Retry_Delay ua="na">3600</Resync_Error_Retry_Delay>
        <Display_Name_1_ ua="na">number</Display_Name_1_>
        <User_ID_1_ ua="na">number</User_ID_1_>
        <Password_1_ ua="na">password</Password_1_>
        <Use_Auth_ID_1_ ua="na">No</Use_Auth_ID_1_>
        <Auth_ID_1_ ua="na"></Auth_ID_1_>
        <Resident_Online_Number_1_ ua="na"></Resident_Online_Number_1_>
        <SIP_URI_1_ ua="na"></SIP_URI_1_>
        <Display_Name_2_ ua="na"></Display_Name_2_>
        <User_ID_2_ ua="na"></User_ID_2_>
        <Password_2_ ua="na"></Password_2_>
        <Use_Auth_ID_2_ ua="na">No</Use_Auth_ID_2_>
        <Auth_ID_2_ ua="na"></Auth_ID_2_>
        <Resident_Online_Number_2_ ua="na"></Resident_Online_Number_2_>
        <SIP_URI_2_ ua="na"></SIP_URI_2_>
    </flat-profile>
    I can see in /admin/config.xml on the adapter that it downloads and loads the settings for only a couple of seconds, but then quickly reverts back to the settings it had before downloading these - what is going on??
    Regards,
    Mikael

    Seems my SPA122 is broken, the same procedure works when I tried another one. Yawn!

  • SPA 122 Loosing Registration

    I'm having the same issue as other posts, and not seeing any solution.
    Hoping that the pieces below narrow down the issue, which I'm feeling is something simple.
    I have 2 numbers registered with voip.ms
    When I was on DSL, no issues the registration --  rarely if ever dropped and usually recovered on its own.
    Now I'm on cable and am getting this nightly drop in registration.
    I have 3 devices and only the SPA122 has an issue with registration.  Motorola TC55 running android and Groundwire - no issues with registration.  Ipad running Softphone - no registration issues.
    When I manually reboot the SPA122, it registers every time.
    I tried setting NAT Keep Alive Msg:  to $REGISTER from the default $NOTIFY, and it hangs in, "fixes" the registration issue ..... but I get no ring through on the SPA122, so that's not a solution.
    I have the logger logging at a "Notice" level, and can send that along -- suffice to say I'm not seeing any error messages.
    Tried various voip.ms servers, no change regarding  the register issue.
    The public IP has been the same same for weeks, so that's not changing.
    All settings are as per Voip.ms wiki, have the latest firmware.
    Help!

    OK thank you Dan for your answer.
    I bought the box from an ISP, but i don't want to use this ISP, so i can not reset the config to default settings without customization ect? With a sample.cfg file from anywhere?

  • SPA 122 stops working with both PHONE 1 and 2 lights flashing

    Hi,
    We have hundreds of units deployed and we start having complaints that the line isn't working anymore while the unit does register with our server. Any calls going to the ATA gets a USER_BUSY condition like if a line was offhook but that's not the case.
    Anyone had this condition? I am running 1.3.1(003).
    One common thing so far is the fact those users do faxing but we ruled out config since we enable T.38 on all of them regardless.
    Since the unit registers, we can't rely on the REGISTRATION too much so would like a way to figure out if the unit is healthy or not.
    Thanks,
    Benoit

    Please update to 1.3.2(014) and see if this resolves the issue.
    http://software.cisco.com/download/release.html?mdfid=283998793&softwareid=282463187&release=1.3.2&relind=AVAILABLE&rellifecycle=&reltype=latest

  • SPA 122 initial config works, but cannot complete outgoing calls

    I configured the ATA exactly step by step anbd I did follow the guidelines of my voip provider (voip.ms).
    I see I am registered, everything works well, I get the dial tone, I can dial but then nothing happens.  I checked the dial plan  and tried several times to reset the unit and configure again from scratch...what am I doing wrong?

    Hello Nathan,
    The configuration of the hunt group is not correct.
    You cannot use call forward all under extension in hunt group because it is ignored this is the reason why the call flow is not working.
    This is one of the restrictions of the hunt groups, please check the following link.
    http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmecover.html#wp1117433
    You need to set the call flow in another way.
    HTH,
    Alex
    *Please rate helpful posts

  • How to record simultaneous datalines each playing in its own thread?

    Hi,
    I have a program where multiple tracks of audio play simultaneously. Each track plays in its own thread to its own sourceDataLine. If I wanted to write all sourceDataLines currently playing to a file, how can I go about this?
    Is there someway to access the central point all of those sourceDataLines are going through and grab the data from there?
    Thanks,
    John

    jmljunior wrote:
    Is there someway to access the central point all of those sourceDataLines are going through and grab the data from there?Ummmm, if your sound card supports loopback capabilities, you could simply record the system mixer and you'd get exactly what was played over the speakers...

  • Can't log into configuration page

    Hi guys,
    i'm new to this forum and seeking help. I just got a new SPA 122 ATA device and i'm trying to loginto the admin page but i keep getting incorrect user/password message.
    i'm typing admin/admin for pass and user and it's not working. I tried to reset the device via IVR menu pressin 73738# but IVR says invalid commmand. I also tried to reset it with the RESET button at the back of the device, but the password still did not work.
    Can someone please help me
    thanks

    Hi Naf,
    You don't mention if you purchased your new SPA122 from an authorized source or if you purchased it from an auction/resale site. The difference is if it was previously deployed by a service provider or skilled end user, they could have locked down the ATA so that you cannot access it without the correct login credentials.
    Some suggestions:
    Just to eliminate variables, make sure you use Internet Explorer to  access the SPA122. There were issues with using other browsers with  earlier SPA122 firmware in the 1.0.x range of firmware
    Make sure that your computer is attached to the ETHERNET port of the SPA122
    The computer should use DHCP and should have received an IP address of 192.168.15.100 from the SPA122Make sure that you are attempting to log in to http://192.168.15.1 from your connected computer. [admin / admin are the correct credentials for a factory default SPA122]
    If all of the above fail and you're able, use Wireshark to determine what files your SPA122 is looking for. Once you know the file name, you can build a configuration file and set the admin user to something known so you can log in again, provided the SPA122 is not completely locked down by the previous user
    Best of luck,
    Patrick---

  • SPA112 Crashes - No dialtone, restart needed

    Yesterday morning, we picked up the phone to make a call via the SPA112 and there was no dialtone.  I disconnected/reconnected the power and then we were able to get a dialtone and make a call.  Unfortunately I didn't have the opportunity to see if the web-interface was up.  Am running the latest GA firmware, 1.1.0 (011) Feb 10 2012.  It had probably been running for only about 4 days or so since the last reboot.  Anyone else seeing stability issues?  (I thought I'd seen another older posting about someone having had to reboot their device regularly.  In addition, I saw the report about caller id's longer than 26 chars causing issues, but I haven't publicized my new number yet so it is unlikely that was the cause.)
    Cisco, any suggestions on a way to capture additional debug info, other than keeping a logging server running?  Didn't see any unusual activity in the device logs.
    Dan

    Hello,
    One thing I know crashes the SPA 122 on FW 1.1.0(011) is a "484 Address Incomplete".  If the box is running 1.0.1(022), it responds appropriately, and just sends reorder to the phone. Of course, that brings back all the issues that were fixed in 1.1.0.
    I opened a case asking about trying the beta to see if it fixes the issues such as the DTMF problem, Caller ID problem, and this issue, but the tech. wanted configs, fresh packet caps, and all kinds of stuff that shouldn't be necessary given that some of the issues I mentioned are supposedly fixed.
    He was real helpful with regard to the config migration from SPA 2102 in that the old config to xml migation only works with a specific version of the SPC tool, which isn't mentioned in the "we're killing SPA2102, but it's OK 'cause you can migate" document.
    Not sure if the 484 box crash is related to the issue discussed here, as it recovers, but one never knows if it causes some progressive issue with the box that after 10 or 20, it burns up all available memory, or something similar. Packet cap and syslog output of the event that crashes the box below.
    23:06:37.954693 IP (tos 0x0, ttl 64, id 22296, offset 0, flags [none], proto UDP (17), length 830)
        10.33.90.60.sip > SPA122.8060: SIP, length: 802
            SIP/2.0 484 Address Incomplete
            Via: SIP/2.0/UDP 10.33.90.111:8060;branch=z9hG4bK-2817ba48
            From: ;tag=1481fc25925927f6o0
            To: ;tag=NNmBD3S5vZSyK
            Call-ID: [email protected]
            CSeq: 102 INVITE
            User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120624T003326Z~86df8b338e+unclean~20120624T044059Z
            Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
            Supported: timer, precondition, path, replaces
            Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
            Reason: Q.850;cause=16;text="NORMAL_CLEARING"
            Content-Length: 0
            Remote-Party-ID: "4669" ;party=calling;privacy=off;screen=no
    23:06:37.973366 IP (tos 0x68, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 635)
        SPA122.8060 > 10.33.90.60.sip: SIP, length: 607
            ACK sip:[email protected] SIP/2.0
            Via: SIP/2.0/UDP 10.33.90.111:8060;branch=z9hG4bK-2817ba48
            From: ;tag=1481fc25925927f6o0
            To: ;tag=NNmBD3S5vZSyK
            Call-ID: [email protected]
            CSeq: 102 ACK
            Max-Forwards: 70
            Proxy-Authorization: Digest username="4331",realm="10.33.90.60",nonce="455f6ec0-c0ce-11e1-bdff-c9d1f242411e",uri="sip:[email protected]",algorithm=MD5,response="e59568b7cab23ca5081d697b612fef0e",qop=auth,nc=00000001,cnonce="be94babd"
            Contact:
            User-Agent: Cisco/SPA122-1.1.0(011)
            Content-Length: 0
    Clock in ATA is slightly diferent than NTP locked server, but death begins at 23:06:37
    <4>Jun 27 23:06:35 SPA122 [17179877.700000]  In cordless Driver Codec 100 and str NSE/8000  chan 0
    10.50.50.1 27/06 23:06:29.268
    <4>Jun 27 23:06:35 SPA122 [17179877.700000]  In cordless Driver Codec 112 and str encaprtp/8000  chan 0
    10.50.50.1 27/06 23:06:29.268
    <29>Jun 27 23:06:37 SPA122 msgswitchd[380]:   MSGSWD RTCP Reqt len 12 Data 2,35,7304,0
    10.50.50.1 27/06 23:06:31.359
    <4>Jun 27 23:06:37 SPA122 [17179879.824000]  RTCP is running so calling rtcp stop
    10.50.50.1 27/06 23:06:31.390
    <4>Jun 27 23:06:37 SPA122 [17179879.824000]  chan->kmode is present not null
    10.50.50.1 27/06 23:06:31.390
    <4>Jun 27 23:06:37 SPA122 [17179879.828000]  ###### RTCP sock_sendmsg return 172
    10.50.50.1 27/06 23:06:31.390
    <4>Jun 27 23:06:37 SPA122 [17179879.836000]  ###### sock_sendmsg return 172
    10.50.50.1 27/06 23:06:31.390
    <4>Jun 27 23:06:37 SPA122 [17179879.836000] 
    10.50.50.1 27/06 23:06:31.452
    <4>Jun 27 23:06:37 SPA122 [17179879.836000]  #### RTP STOP Flag set in this channel break ####
    10.50.50.1 27/06 23:06:31.452
    <6>Jun 27 23:06:37 SPA122 [17179879.888000] cordless: deinit
    10.50.50.1 27/06 23:06:31.452
    <6>Jun 27 23:06:52 SPA122 [17179894.068000] cordless: loading synergy-2012-01-18
    10.50.50.1 27/06 23:06:45.757
    <6>Jun 27 23:06:52 SPA122 [17179894.108000] cordless: init successful
    10.50.50.1 27/06 23:06:45.757
    <13>Jun 27 23:06:54 SPA122 msgswitchd:  MSGSWITCH fd_rtp fifo created 12
    10.50.50.1 27/06 23:06:47.817
    <13>Jun 27 23:06:54 SPA122 msgswitchd:  MSGSWITCH fd_ch fifo created 14
    10.50.50.1 27/06 23:06:47.817
    <13>Jun 27 23:06:54 SPA122 msgswitchd:  MSGSWITCH fd_gmep fifo created 15
    10.50.50.1 27/06 23:06:47.817

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