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)