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IVR in coming calls

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gmr

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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.
 
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.
 
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:[email protected]:5060 has failed; Cause: 487 Request Cancelled; from IP:172.16.1.87:5060
13:35:30.875 [CM503003]: Call(80): Call to sip:[email protected]:5060 has failed; Cause: 487 Request Cancelled; from IP:172.16.1.91:5060
13:35:30.859 [CM503003]: Call(80): Call to sip:[email protected] has failed; Cause: 487 Request Cancelled; from IP:172.16.1.85:5060
13:35:30.843 [CM503003]: Call(80): Call to sip:[email protected] has failed; Cause: 487 Request Cancelled; from IP:172.16.1.63:5060
13:35:30.843 [CM503003]: Call(80): Call to sip:[email protected] 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:[email protected]:40600;rinstance=57265a2c5f039023]
13:35:30.734 [CM503005]: Call(80): Forwarding: Ext:Ext.801@[Dev:sip:[email protected]:40600;rinstance=57265a2c5f039023]
13:35:28.375 [CM102001]: Authentication failed for SipReq: INVITE [email protected]:5060 tid=5b4141b386d298a2 cseq=INVITE [email protected]: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 [email protected]:5060 tid=f3800e903dc7cb50 cseq=INVITE [email protected]: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:[email protected]:5060]
13:35:00.968 [CM503002]: Call(80): Alerting sip:[email protected]: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:[email protected]:5060]
13:35:00.921 [CM503002]: Call(80): Alerting sip:[email protected]: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:[email protected]:5060]
13:35:00.921 [CM503002]: Call(80): Alerting sip:[email protected]: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:[email protected]:5060]
13:35:00.796 [CM503002]: Call(80): Alerting sip:[email protected]: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:[email protected]:5060]
13:35:00.796 [CM503002]: Call(80): Alerting sip:[email protected]: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:[email protected]:5060]
13:35:00.796 [CM503002]: Call(80): Alerting sip:[email protected]: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:[email protected]:5060]
13:35:00.781 [CM503002]: Call(80): Alerting sip:[email protected]: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:[email protected]:5060]
13:35:00.781 [CM503002]: Call(80): Alerting sip:[email protected]: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:[email protected]: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:[email protected]: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:[email protected]: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:[email protected]:5060]
13:35:00.718 [CM503025]: Call(80): Calling RingAll800:100Ext.100101Ext.101102Ext.102103Ext.103104Ext.104105Ext.105106Ext.106107Ext.107@[Dev:sip:[email protected]:5060]
13:35:00.718 [CM503025]: Call(80): Calling RingAll800:100Ext.100101Ext.101102Ext.102103Ext.103104Ext.104105Ext.105106Ext.106107Ext.107@[Dev:sip:[email protected]: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:[email protected]:5060]
13:35:00.703 [CM503025]: Call(80): Calling RingAll800:100Ext.100101Ext.101102Ext.102103Ext.103104Ext.104105Ext.105106Ext.106107Ext.107@[Dev:sip:[email protected]: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:[email protected]:5060;transport=udp;user=phone,Dev:sip:[email protected]:5060,Dev:sip:[email protected]:5060;transport=udp;user=phone,Dev:sip:[email protected]:5060,Dev:sip:[email protected]:5060,Dev:sip:[email protected]:5060;transport=udp;user=phone,Dev:sip:[email protected]:5060;transport=udp;user=phone,Dev:sip:[email protected]:5060;transport=udp;user=phone]
13:35:00.656 [CM503010]: Making route(s) to <sip:[email protected]:5060>
13:35:00.656 Refer: from=<sip:[email protected]:5060>;tag=1b51f20f; to="PCHC Ring Group:WIRELESS CALLER"<sip:[email protected]:5060;nf=e>;tag=66396f70; RefTo=<sip:[email protected]: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)
 
Hi, if you would like i can login and take a look for you?
 
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
Click to expand...

Well the below snippet from the logs you posted show otherwise.

13:35:28.375 [CM102001]: Authentication failed for SipReq: INVITE [email protected]:5060 tid=5b4141b386d298a2 cseq=INVITE [email protected]: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
 
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
 
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