IVR in coming calls

Discussion in '3CX Phone System - General' started by gmr, Jul 20, 2011.

Thread Status:
Not open for further replies.
  1. gmr

    gmr

    Joined:
    May 18, 2011
    Messages:
    10
    Likes Received:
    0
    I use a GWX4108 with 6 incoming PTSN lines. Can the IVR answer more than incoming call at a time? Our experienced is that when the digitalreceptionist ls handling a calll, other incoming calls ring and are not processed until the first call is fully processed.. This has caused alot of our customers to think we are not open and/or not answering the phone.
     
  2. davidbenwell

    davidbenwell Active Member

    Joined:
    Apr 27, 2010
    Messages:
    704
    Likes Received:
    0
    the IVR can handle any number of calls at once up to the 3CX Licence Limit.

    what you need to do is make two calls at once so its logged in the 3CX logs and then post the logs here for us to see.
     
    Stop hovering to collapse... Click to collapse... Hover to expand... Click to expand...
  3. gmr

    gmr

    Joined:
    May 18, 2011
    Messages:
    10
    Likes Received:
    0
    Below is log from two simultaneous calls. Active call 10000 2nd call 10001. Have checked auth ID and passwords and they do appear to match settings in extention. Still unable to process two incoming calls at once.8 seat licence on Server 2003 R2 GWX 4108

    13:35:30.890 [CM503003]: Call(80): Call to sip:107@172.16.1.17:5060 has failed; Cause: 487 Request Cancelled; from IP:172.16.1.87:5060
    13:35:30.875 [CM503003]: Call(80): Call to sip:105@172.16.1.17:5060 has failed; Cause: 487 Request Cancelled; from IP:172.16.1.91:5060
    13:35:30.859 [CM503003]: Call(80): Call to sip:104@172.16.1.17 has failed; Cause: 487 Request Cancelled; from IP:172.16.1.85:5060
    13:35:30.843 [CM503003]: Call(80): Call to sip:101@172.16.1.17 has failed; Cause: 487 Request Cancelled; from IP:172.16.1.63:5060
    13:35:30.843 [CM503003]: Call(80): Call to sip:103@172.16.1.17 has failed; Cause: 487 Request Cancelled; from IP:172.16.1.83:5060
    13:35:30.781 [CM503025]: Call(80): Calling Ext:Ext.801@[Dev:sip:801@127.0.0.1:40600;rinstance=57265a2c5f039023]
    13:35:30.734 [CM503005]: Call(80): Forwarding: Ext:Ext.801@[Dev:sip:801@127.0.0.1:40600;rinstance=57265a2c5f039023]
    13:35:28.375 [CM102001]: Authentication failed for SipReq: INVITE 10000@172.16.1.17:5060 tid=5b4141b386d298a2 cseq=INVITE contact=10001@172.16.1.4:5062 / 20217 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
    13:35:16.203 [CM102001]: Authentication failed for SipReq: INVITE 10000@172.16.1.17:5060 tid=f3800e903dc7cb50 cseq=INVITE contact=10001@172.16.1.4:5062 / 12790 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
    13:35:07.671 Currently active calls - 1: [80]
    13:35:00.968 [CM505001]: Ext.103: Device info: Device Identified: [Man: Polycom;Mod: SoundPoint IP series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [PolycomSoundPointIP-SPIP_650-UA/3.0.4.0061] PBX contact: [sip:103@172.16.1.17:5060]
    13:35:00.968 [CM503002]: Call(80): Alerting sip:103@172.16.1.83:5060
    13:35:00.921 [CM505001]: Ext.104: Device info: Device Identified: [Man: Polycom;Mod: SoundPoint IP series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [PolycomSoundPointIP-SPIP_650-UA/3.0.4.0061] PBX contact: [sip:104@172.16.1.17:5060]
    13:35:00.921 [CM503002]: Call(80): Alerting sip:104@172.16.1.85:5060
    13:35:00.921 [CM505001]: Ext.101: Device info: Device Identified: [Man: Polycom;Mod: SoundPoint IP series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [PolycomSoundPointIP-SPIP_650-UA/3.0.4.0061] PBX contact: [sip:101@172.16.1.17:5060]
    13:35:00.921 [CM503002]: Call(80): Alerting sip:101@172.16.1.63:5060
    13:35:00.796 [CM505001]: Ext.100: Device info: Device Identified: [Man: GrandStream;Mod: GXP series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXP2020 1.2.5.3] PBX contact: [sip:100@172.16.1.17:5060]
    13:35:00.796 [CM503002]: Call(80): Alerting sip:100@172.16.1.95:5060;transport=udp;user=phone
    13:35:00.796 [CM505001]: Ext.106: Device info: Device Identified: [Man: GrandStream;Mod: GXP series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXP2020 1.2.5.3] PBX contact: [sip:106@172.16.1.17:5060]
    13:35:00.796 [CM503002]: Call(80): Alerting sip:106@172.16.1.89:5060;transport=udp;user=phone
    13:35:00.796 [CM505001]: Ext.107: Device info: Device Identified: [Man: GrandStream;Mod: GXP series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXP2020 1.2.5.3] PBX contact: [sip:107@172.16.1.17:5060]
    13:35:00.796 [CM503002]: Call(80): Alerting sip:107@172.16.1.87:5060;transport=udp;user=phone
    13:35:00.781 [CM505001]: Ext.105: Device info: Device Identified: [Man: GrandStream;Mod: GXP series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXP2020 1.2.5.3] PBX contact: [sip:105@172.16.1.17:5060]
    13:35:00.781 [CM503002]: Call(80): Alerting sip:105@172.16.1.91:5060;transport=udp;user=phone
    13:35:00.781 [CM505001]: Ext.102: Device info: Device Identified: [Man: GrandStream;Mod: GXP series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXP2020 1.2.5.3] PBX contact: [sip:102@172.16.1.17:5060]
    13:35:00.781 [CM503002]: Call(80): Alerting sip:102@172.16.1.81:5060;transport=udp;user=phone
    13:35:00.734 [CM503025]: Call(80): Calling RingAll800:100Ext.100101Ext.101102Ext.102103Ext.103104Ext.104105Ext.105106Ext.106107Ext.107@[Dev:sip:107@172.16.1.87:5060;transport=udp;user=phone]
    13:35:00.734 [CM503025]: Call(80): Calling RingAll800:100Ext.100101Ext.101102Ext.102103Ext.103104Ext.104105Ext.105106Ext.106107Ext.107@[Dev:sip:106@172.16.1.89:5060;transport=udp;user=phone]
    13:35:00.734 [CM503025]: Call(80): Calling RingAll800:100Ext.100101Ext.101102Ext.102103Ext.103104Ext.104105Ext.105106Ext.106107Ext.107@[Dev:sip:105@172.16.1.91:5060;transport=udp;user=phone]
    13:35:00.718 [CM503025]: Call(80): Calling RingAll800:100Ext.100101Ext.101102Ext.102103Ext.103104Ext.104105Ext.105106Ext.106107Ext.107@[Dev:sip:104@172.16.1.85:5060]
    13:35:00.718 [CM503025]: Call(80): Calling RingAll800:100Ext.100101Ext.101102Ext.102103Ext.103104Ext.104105Ext.105106Ext.106107Ext.107@[Dev:sip:103@172.16.1.83:5060]
    13:35:00.718 [CM503025]: Call(80): Calling RingAll800:100Ext.100101Ext.101102Ext.102103Ext.103104Ext.104105Ext.105106Ext.106107Ext.107@[Dev:sip:102@172.16.1.81:5060;transport=udp;user=phone]
    13:35:00.703 [CM503025]: Call(80): Calling RingAll800:100Ext.100101Ext.101102Ext.102103Ext.103104Ext.104105Ext.105106Ext.106107Ext.107@[Dev:sip:101@172.16.1.63:5060]
    13:35:00.703 [CM503025]: Call(80): Calling RingAll800:100Ext.100101Ext.101102Ext.102103Ext.103104Ext.104105Ext.105106Ext.106107Ext.107@[Dev:sip:100@172.16.1.95:5060;transport=udp;user=phone]
    13:35:00.656 [CM503004]: Call(80): Route 1: RingAll800:100Ext.100101Ext.101102Ext.102103Ext.103104Ext.104105Ext.105106Ext.106107Ext.107@[Dev:sip:100@172.16.1.95:5060;transport=udp;user=phone,Dev:sip:101@172.16.1.63:5060,Dev:sip:102@172.16.1.81:5060;transport=udp;user=phone,Dev:sip:103@172.16.1.83:5060,Dev:sip:104@172.16.1.85:5060,Dev:sip:105@172.16.1.91:5060;transport=udp;user=phone,Dev:sip:106@172.16.1.89:5060;transport=udp;user=phone,Dev:sip:107@172.16.1.87:5060;transport=udp;user=phone]
    13:35:00.656 [CM503010]: Making route(s) to <sip:800@127.0.0.1:5060>
    13:35:00.656 Refer: from=<sip:801@127.0.0.1:5060>;tag=1b51f20f; to="PCHC Ring Group:WIRELESS CALLER"<sip:8657555171@127.0.0.1:5060;nf=e>;tag=66396f70; RefTo=<sip:800@127.0.0.1:5060>


    Phone Number Settings
    Channel(s) SIP User ID Authenticate ID Authen Password Profile ID
    1. Profile 1 Profile 2 Profile 3
    2. Profile 1 Profile 2 Profile 3
    3. Profile 1 Profile 2 Profile 3
    4. Profile 1 Profile 2 Profile 3
    5. Profile 1 Profile 2 Profile 3
    6. Profile 1 Profile 2 Profile 3
    7. Profile 1 Profile 2 Profile 3
    8. Profile 1 Profile 2 Profile 3


    Call Progress Tones
    [Syntax: ch x-y: f1=val@vol,f2=val@vol,c=on1/off1-on2/off2-on3/off3; ...]
    Note : f1,f2-frequency(Hz); vol-volume(dB); c-cadence(10ms, 0-continuous)
    1. Dial Tone:
    2. Ringback Tone:
    3. Busy Tone:
    4. Reorder Tone:


    Channel Voice Setting
    1. Tx to PSTN Audio Gain(dB): (-12-12, default 1)
    2. Rx from PSTN Audio Gain(dB): (-12-12, default 0)
    3. Silence Suppression(Y/N): (default Yes)
    4. Echo Cancellation(Y/N): (default Yes)


    Channel Specific Setting
    1. DTMF Methods(1-7): (default 1)
    (1:in-audio, 2:RFC2833, 3:1+2, 4:SIP Info, 5:1+4, 6:2+4, 7:1+2+4)
    2. No Key Entry Timeout(X1s): (1-9, default 4)
    3. Local SIP Listen Port: (default ch1-8:5060++;)
    4. SRTP Mode(1-3): (default 1)
    (1:disabled, 2:enabled but not forced, 3:enabled and forced)


    Port Scheduling Schema (Voip->PSTN)
    1. Round-robin and/or Flexible: (default rr:1-8;)
    (Syntax: rr: port_group; [...])
    (Default: rr:1-8; round-robin of all ports )
    2. Prefix to Specify Port(1 stage dialing method): (default 99)
    (Syntax: prefix# + ch# + dialing# will request the ch# per call)
    (Note that this code has to prefix dialplan number and prefix doesn't impact round-robin)
     
  4. davidbenwell

    davidbenwell Active Member

    Joined:
    Apr 27, 2010
    Messages:
    704
    Likes Received:
    0
    Hi, if you would like i can login and take a look for you?
     
    Stop hovering to collapse... Click to collapse... Hover to expand... Click to expand...
  5. 3cx

    3cx

    Joined:
    Mar 24, 2008
    Messages:
    24
    Likes Received:
    0
    Well the below snippet from the logs you posted show otherwise.

    13:35:28.375 [CM102001]: Authentication failed for SipReq: INVITE 10000@172.16.1.17:5060 tid=5b4141b386d298a2 cseq=INVITE contact=10001@172.16.1.4:5062 / 20217 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings

    This shows that an invite from 10000 is trying to go over line 10001 so this becomes an authentication issue.

    You might have configured the gateway incorrectly. follow this guide please
    http: //www.3cx.com/voip-gateways/Grandstream-GXW-41044108/ NO LONGER AVAILABLE
     
  6. gmr

    gmr

    Joined:
    May 18, 2011
    Messages:
    10
    Likes Received:
    0
    Problem appears to be solved. SIP ID, Auth ID and pasword were OK. Needed to reconfigure Call Progress Tones from default setting GXW 4108 to specfic US settings. Refer to: How do I configure Tonesets for my Country on a Grandstream GXW-4104 / GXW-4108? Found in: Home » Docs & FAQ

    Thanks for input
     
Thread Status:
Not open for further replies.