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d1rage5

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Posts posted by d1rage5

    ivm cc payment

    in IVM

    Posted

    Yes. You can have the call answered by any OGM you wish and then after storing the person's name and address to a log/database, use the OGM action to send them to Voice Mail to leave a message.

     

    so after the last section (Done) that will play back there info and they press 1 to confirm you would then change it to an OGM that leaves a message? Would it still finish and record the data in the DB/CVS?

    ivm cc payment

    in IVM

    Posted

    i would like to know if you could if use this to allow the person calling to leave a voice mail as well so that you have the name and address and then have the data entered?

    Blacklisted phone numbers

    in Axon

    Posted

    Axon does not support this feature natively.. however I think there may be a way to accomplish what you are trying to do. If you had a softphone set-up as an extension that could check the caller ID against a blacklist and then quickly answer and hang-up if the number is on the list... then any incoming call on that ring group would also ring the soft phone and if the number had been black listed the call would immediately be answered and disconnected. If you wanted, you could add a delay of 1 second to the other extensions in the group so that a black-listed caller wouldn't cause the other phones to ring at all (where as without a delay, a black-listed caller would still ring once).

     

    The problem is, I do not know of any soft phone with that type of black-listing capability.

     

     

    if you are using the IVM you can block nubers with CID.

    goto settings then caller id the enable the call id block and put the numbers you want to block or you can enable the blocking of private or unknow numbers.

     

    you have options to send them to a OGM or just drop the call.

    Music on Hold with SPA942 and Axon

    in Axon

    Posted

    Ive also seen on a lot of home routers and firewall the stun servers will screw things up, you can try not to use then in axon and the MOH (this will have to be done on the local host file and have it point to the loopback address). after this i was able to put the 197 by itself on the MOH tab and it allowed me to place people on hold.

    external line cant delete

    in Axon

    Posted

    I created some external lines and the have like 25 charters with either & sign and numbers but they can not be deleted, edited or changed. every time i try to edit them it will just create a new line (that new one can be deleted as long as it is under 20 charters).

    has any one else seen this and know how to correct or manully delete the entry?

    grandstream gxp2000 and MOH

    in IMS

    Posted

    i have had several people say that their gxp2000 phone works fine with the IMS but they can not tell me hows its configured to work.

    I will post below the data from the gxp2000 to the PBX server that also holds the IVM and IMS, no where on the phone can you set what port is used for MOH system (port 606) unles it is being configured some other way to get it to work, this is also a problem on most other ip phone except the stupid softphone. I can make and receive calls all day long but when placeing someone on hold they get nothing even internally. If i dial the 197 ext i can access the MOH.....

     

     

    heres the log from gxp2000 to Axon and then when i put someone on hold.. even someone internal....

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:04:51 LCD Callmode: CALLMODE_SPEAKERPHONE 10.33.15.155 29/03 21:05:45.832

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:04:51 Voc mode (0): CALLMODE_SPEAKERPHONE 10.33.15.155 29/03 21:05:45.832

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:04:51 Aud path (0): AUD_PATH_HANDSFREE 10.33.15.155 29/03 21:05:45.832

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Stop RTP Keep-alive: Channel 0 lport 0 10.33.15.155 29/03 21:05:45.847

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: INVITE To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:05:45.863

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:04:51 [status]-OFF HOOK 10.33.15.155 29/03 21:05:45.910

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIPReceive(770, Account1): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bK0758995857fac462;rport To: <sip:816783136925@10.33.15.45>;tag=7360 From: "Daniel Murray" <sip:105@10.33.15.4 10.33.15.155 29/03 21:05:45.957

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Received SIP message: 200 10.33.15.155 29/03 21:05:45.972

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIP dialog matched to channel 0 10.33.15.155 29/03 21:05:45.972

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:04:51 Tone stop (0) 10.33.15.155 29/03 21:05:45.972

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Session Info: Payload-Type=0, Frames/Packet=3, DTMF=101 10.33.15.155 29/03 21:05:45.972

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: ACK To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:05:45.972

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] sip_len: 716, sip_handle: 0x0047F0AA, ACK sip:816783136925@10.33.15.45:5060 SIP/2.0 Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bKb95f4e8f01bf2868;rport From: "Daniel Murray" <sip:105@10.33.15.45>; 10.33.15.155 29/03 21:05:46.003

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] RTP session starts. Channel: 0 Local RTP port: 5004 Remote RTP endpoint: 10.33.15.45:8002 10.33.15.155 29/03 21:05:46.003

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:04:52 Tone stop (0) 10.33.15.155 29/03 21:05:46.066

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: INVITE To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:05:50.441

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Start RTP Keep-alive: Channel 0 lport 0 account 0 stun 1 keep_alive 1 interval 20 10.33.15.155 29/03 21:05:50.504

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:04:56 [status]-ON HOOK 10.33.15.155 29/03 21:05:50.519

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:04:56 Tone stop (0) 10.33.15.155 29/03 21:05:50.519

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:04:56 LCD Callmode: CALLMODE_NULL 10.33.15.155 29/03 21:05:50.519

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:04:56 Voc mode (0): CALLMODE_NULL 10.33.15.155 29/03 21:05:50.535

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:04:56 Aud path (0): AUD_PATH_NULL 10.33.15.155 29/03 21:05:50.535

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIPReceive(841, Account1): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bKe2e704ae609a7b13;rport To: <sip:816783136925@10.33.15.45>;tag=7360 From: "Daniel Murray" <sip:105@10.33.15.4 10.33.15.155 29/03 21:05:50.566

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Received SIP message: 200 10.33.15.155 29/03 21:05:50.566

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIP dialog matched to channel 0 10.33.15.155 29/03 21:05:50.582

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:04:56 Tone stop (0) 10.33.15.155 29/03 21:05:50.582

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Session Info: Payload-Type=0, Frames/Packet=3, DTMF=101 10.33.15.155 29/03 21:05:50.582

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: ACK To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:05:50.582

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] sip_len: 716, sip_handle: 0x0047F0AA, ACK sip:816783136925@10.33.15.45:5060 SIP/2.0 Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bK6b7078a077502887;rport From: "Daniel Murray" <sip:105@10.33.15.45>; 10.33.15.155 29/03 21:05:50.598

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:05:15 LCD Callmode: CALLMODE_SPEAKERPHONE 10.33.15.155 29/03 21:06:09.521

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:05:15 Voc mode (0): CALLMODE_SPEAKERPHONE 10.33.15.155 29/03 21:06:09.521

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:05:15 Aud path (0): AUD_PATH_HANDSFREE 10.33.15.155 29/03 21:06:09.521

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Stop RTP Keep-alive: Channel 0 lport 0 10.33.15.155 29/03 21:06:09.521

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: INVITE To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:06:09.552

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:05:15 [status]-OFF HOOK 10.33.15.155 29/03 21:06:09.599

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIPReceive(770, Account1): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bKdd89c65cf95c18fe;rport To: <sip:816783136925@10.33.15.45>;tag=7360 From: "Daniel Murray" <sip:105@10.33.15.4 10.33.15.155 29/03 21:06:09.630

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Received SIP message: 200 10.33.15.155 29/03 21:06:09.646

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIP dialog matched to channel 0 10.33.15.155 29/03 21:06:09.646

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:05:15 Tone stop (0) 10.33.15.155 29/03 21:06:09.661

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Session Info: Payload-Type=0, Frames/Packet=3, DTMF=101 10.33.15.155 29/03 21:06:09.661

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: ACK To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:06:09.661

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] sip_len: 716, sip_handle: 0x0047F0AA, ACK sip:816783136925@10.33.15.45:5060 SIP/2.0 Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bK04af718f45df7769;rport From: "Daniel Murray" <sip:105@10.33.15.45>; 10.33.15.155 29/03 21:06:09.677

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] RTP session starts. Channel: 0 Local RTP port: 5004 Remote RTP endpoint: 10.33.15.45:8002 10.33.15.155 29/03 21:06:09.693

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:05:15 Tone stop (0) 10.33.15.155 29/03 21:06:09.740

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: INVITE To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:06:38.742

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Start RTP Keep-alive: Channel 0 lport 0 account 0 stun 1 keep_alive 1 interval 20 10.33.15.155 29/03 21:06:38.804

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:05:44 [status]-ON HOOK 10.33.15.155 29/03 21:06:38.804

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:05:44 Tone stop (0) 10.33.15.155 29/03 21:06:38.820

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:05:44 LCD Callmode: CALLMODE_NULL 10.33.15.155 29/03 21:06:38.820

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:05:44 Voc mode (0): CALLMODE_NULL 10.33.15.155 29/03 21:06:38.820

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:05:44 Aud path (0): AUD_PATH_NULL 10.33.15.155 29/03 21:06:38.820

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIPReceive(841, Account1): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bK31dc9846390d95f8;rport To: <sip:816783136925@10.33.15.45>;tag=7360 From: "Daniel Murray" <sip:105@10.33.15.4 10.33.15.155 29/03 21:06:38.851

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Received SIP message: 200 10.33.15.155 29/03 21:06:38.867

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIP dialog matched to channel 0 10.33.15.155 29/03 21:06:38.867

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:05:44 Tone stop (0) 10.33.15.155 29/03 21:06:38.867

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Session Info: Payload-Type=0, Frames/Packet=3, DTMF=101 10.33.15.155 29/03 21:06:38.867

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: ACK To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:06:38.867

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] sip_len: 716, sip_handle: 0x0047F0AA, ACK sip:816783136925@10.33.15.45:5060 SIP/2.0 Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bK9172ca411d7184c0;rport From: "Daniel Murray" <sip:105@10.33.15.45>; 10.33.15.155 29/03 21:06:38.898

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] RTPSendKeepAlive: Channel 0 lport 0 10.33.15.155 29/03 21:06:58.790

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:06:05 LCD Callmode: CALLMODE_SPEAKERPHONE 10.33.15.155 29/03 21:06:59.462

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:06:05 Voc mode (0): CALLMODE_SPEAKERPHONE 10.33.15.155 29/03 21:06:59.462

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:06:05 Aud path (0): AUD_PATH_HANDSFREE 10.33.15.155 29/03 21:06:59.462

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Stop RTP Keep-alive: Channel 0 lport 0 10.33.15.155 29/03 21:06:59.462

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: INVITE To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:06:59.493

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:06:05 [status]-OFF HOOK 10.33.15.155 29/03 21:06:59.540

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIPReceive(770, Account1): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bKfbc27c750448d548;rport To: <sip:816783136925@10.33.15.45>;tag=7360 From: "Daniel Murray" <sip:105@10.33.15.4 10.33.15.155 29/03 21:06:59.571

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Received SIP message: 200 10.33.15.155 29/03 21:06:59.587

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIP dialog matched to channel 0 10.33.15.155 29/03 21:06:59.587

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:06:05 Tone stop (0) 10.33.15.155 29/03 21:06:59.587

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Session Info: Payload-Type=0, Frames/Packet=3, DTMF=101 10.33.15.155 29/03 21:06:59.602

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: ACK To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:06:59.602

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] sip_len: 716, sip_handle: 0x0047F0AA, ACK sip:816783136925@10.33.15.45:5060 SIP/2.0 Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bK5a8805ef1d9f28df;rport From: "Daniel Murray" <sip:105@10.33.15.45>; 10.33.15.155 29/03 21:06:59.618

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] RTP session starts. Channel: 0 Local RTP port: 5004 Remote RTP endpoint: 10.33.15.45:8002 10.33.15.155 29/03 21:06:59.634

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:06:05 Tone stop (0) 10.33.15.155 29/03 21:06:59.681

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: INVITE To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:07:13.963

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Start RTP Keep-alive: Channel 0 lport 0 account 0 stun 1 keep_alive 1 interval 20 10.33.15.155 29/03 21:07:14.025

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:06:19 [status]-ON HOOK 10.33.15.155 29/03 21:07:14.041

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:06:19 Tone stop (0) 10.33.15.155 29/03 21:07:14.041

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:06:19 LCD Callmode: CALLMODE_NULL 10.33.15.155 29/03 21:07:14.041

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:06:19 Voc mode (0): CALLMODE_NULL 10.33.15.155 29/03 21:07:14.041

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:06:19 Aud path (0): AUD_PATH_NULL 10.33.15.155 29/03 21:07:14.041

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIPReceive(841, Account1): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bK35f2c5ee3d580f1f;rport To: <sip:816783136925@10.33.15.45>;tag=7360 From: "Daniel Murray" <sip:105@10.33.15.4 10.33.15.155 29/03 21:07:14.088

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Received SIP message: 200 10.33.15.155 29/03 21:07:14.088

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIP dialog matched to channel 0 10.33.15.155 29/03 21:07:14.088

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:06:19 Tone stop (0) 10.33.15.155 29/03 21:07:14.103

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Session Info: Payload-Type=0, Frames/Packet=3, DTMF=101 10.33.15.155 29/03 21:07:14.103

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: ACK To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:07:14.103

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] sip_len: 716, sip_handle: 0x0047F0AA, ACK sip:816783136925@10.33.15.45:5060 SIP/2.0 Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bK78d91c826561d791;rport From: "Daniel Murray" <sip:105@10.33.15.45>; 10.33.15.155 29/03 21:07:14.119

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] RTPSendKeepAlive: Channel 0 lport 0 10.33.15.155 29/03 21:07:34.027

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] RTPSendKeepAlive: Channel 0 lport 0 10.33.15.155 29/03 21:07:54.028

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] RTPSendKeepAlive: Channel 0 lport 0 10.33.15.155 29/03 21:08:14.030

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:21 [status]-OFF HOOK 10.33.15.155 29/03 21:08:15.530

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:21 LCD Callmode: CALLMODE_HANDSET 10.33.15.155 29/03 21:08:15.545

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:21 Voc mode (0): CALLMODE_HANDSET 10.33.15.155 29/03 21:08:15.545

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:21 Aud path (0): AUD_PATH_HANDSET 10.33.15.155 29/03 21:08:15.545

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:21 Tone start (0): 66 10.33.15.155 29/03 21:08:15.545

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:21 [status]-ON HOOK 10.33.15.155 29/03 21:08:15.858

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:21 Tone stop (0) 10.33.15.155 29/03 21:08:15.873

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:21 LCD Callmode: CALLMODE_NULL 10.33.15.155 29/03 21:08:15.873

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:21 Voc mode (0): CALLMODE_NULL 10.33.15.155 29/03 21:08:15.873

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:21 Aud path (0): AUD_PATH_NULL 10.33.15.155 29/03 21:08:15.873

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:25 LCD Callmode: CALLMODE_SPEAKERPHONE 10.33.15.155 29/03 21:08:19.999

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:25 Voc mode (0): CALLMODE_SPEAKERPHONE 10.33.15.155 29/03 21:08:19.999

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:25 Aud path (0): AUD_PATH_HANDSFREE 10.33.15.155 29/03 21:08:19.999

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Stop RTP Keep-alive: Channel 0 lport 0 10.33.15.155 29/03 21:08:19.999

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: INVITE To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:08:20.030

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:26 [status]-OFF HOOK 10.33.15.155 29/03 21:08:20.092

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIPReceive(770, Account1): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bK5a3144786cf9d7f9;rport To: <sip:816783136925@10.33.15.45>;tag=7360 From: "Daniel Murray" <sip:105@10.33.15.4 10.33.15.155 29/03 21:08:20.139

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Received SIP message: 200 10.33.15.155 29/03 21:08:20.155

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIP dialog matched to channel 0 10.33.15.155 29/03 21:08:20.155

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:26 Tone stop (0) 10.33.15.155 29/03 21:08:20.155

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Session Info: Payload-Type=0, Frames/Packet=3, DTMF=101 10.33.15.155 29/03 21:08:20.155

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: ACK To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:08:20.155

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] sip_len: 716, sip_handle: 0x0047F0AA, ACK sip:816783136925@10.33.15.45:5060 SIP/2.0 Via: SIP/2.0/UDP 10.33.15.155:5060;branch=z9hG4bKa2310b21adc4c885;rport From: "Daniel Murray" <sip:105@10.33.15.45>; 10.33.15.155 29/03 21:08:20.186

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] RTP session starts. Channel: 0 Local RTP port: 5004 Remote RTP endpoint: 10.33.15.45:8002 10.33.15.155 29/03 21:08:20.186

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:26 Tone stop (0) 10.33.15.155 29/03 21:08:20.217

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIPReceive(367, Account1): BYE sip:105@10.33.15.155;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.33.15.45:5060;rport;branch=z9hG4bK1861120 To: "Daniel Murray" <sip:105@10.33.15.45>;tag=a2252fe7a7e7cb6a 10.33.15.155 29/03 21:08:28.968

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Received SIP message: 4 10.33.15.155 29/03 21:08:28.968

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] SIP dialog matched to channel 0 10.33.15.155 29/03 21:08:28.984

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] Send SIP message: 200 To 10.33.15.45:5060, sip_handle: 0x0047F0AA 10.33.15.155 29/03 21:08:28.984

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] sip_len: 536, sip_handle: 0x0047F0AA, SIP/2.0 200 OK Via: SIP/2.0/UDP 10.33.15.45:5060;rport;branch=z9hG4bK1861120 From: <sip:816783136925@10.33.15.45>;tag=7360 To: "Daniel Murray" <sip:105@10.33.15. 10.33.15.155 29/03 21:08:29.015

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:34 LCD Callmode: CALLMODE_NULL 10.33.15.155 29/03 21:08:29.015

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:34 Voc mode (0): CALLMODE_NULL 10.33.15.155 29/03 21:08:29.015

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:34 Aud path (0): AUD_PATH_NULL 10.33.15.155 29/03 21:08:29.015

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] sending notifies from sig_remote_dc 10.33.15.155 29/03 21:08:29.031

    <14>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:34 [status]-ON HOOK 10.33.15.155 29/03 21:08:29.031

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:34 Tone stop (0) 10.33.15.155 29/03 21:08:29.031

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:34 LCD Callmode: CALLMODE_NULL 10.33.15.155 29/03 21:08:29.031

    <15>GS_LOG: [00:0B:82:0A:3C:2F][000][FF71][01010610] 2008-03-29 21:07:34 Voc mode (0): CALLMODE_NULL 10.33.15.155 29/03 21:08:29.031

    MOH being blocked on users home router

    in IMS

    Posted

    we have axon setup on a firewall at a corp location the firewall has a static IP and forwards all request to axon server with IVR and IMS (using port 606) the following is what the user gets on there router when placing someone on hold.

     

    this is a d-link dsl router

     

    Blocked incoming ICMP error message (ICMP type 3) from X.X.X.237 (Corp static ip on firewall) to X.X.X.X (static IP on users router) as there is no UDP session active between X.X.X.X:5060 (users Static IP on router) and 10.30.32.45:606 (private ip on corp server behind firewall.

     

    X.X.X.X are public static IPs

    MOH plays for a second before IVM OGM plays

    in IVM

    Posted

    I got it to work (at least a work around), under groups in AXON change the on hold while waiting to ring only.

     

    this will ring then ivm ANSWERS (ONLY ONE RING FROM AXON) this still needs a better fix, hope someone else will chime in with the correct way to do it.

     

     

    For those who would like to know, it was a memory problem changed out the 4 gig (had two bad sticks on each pair) just goes to show how wird memory issues will drive you crazy

    music on hold picks up before IVM in axon

    in Axon

    Posted

    Well i know that ext groups were made for group ringing as we are going to use this we just had one person and adding others next week). We have been testing it setup this way without issue on the test server (much faster system), we had a few other odd ball things happening with axon.

     

    SO i thought may be some flacky memory and would you know that fixed all the problems - HP will love me sending back 4 gig of ram.....

     

    Thanks

    skype call to SIP (AXON)

    in Uplink Skype to SIP Adapter

    Posted

    when a skype caller calls in the uplink passes the call to axon then axon passes the call to IVM but the skype user can not dial the ext needed. it does not pass the tone used to transfer calls and they will just get voice mail after listening to the OGM 3 times.

     

    has anyone seen how to correct this?

    call transfer to outside line by user

    in IVM

    Posted

    This may need to be an axon add or ivm (as this is were it is now) give the user the option to add a number to transfer a call to (like there cell phone or other ext) i would think it could be added to the menu when they check there messages (199).

     

    Also give them a option to setup a vacation OGM insteed of just one that they need to change every time.

    music on hold picks up before IVM in axon

    in Axon

    Posted

    in axon you have the group line to pick up calls on, in the option for waiting to transfer it gives three options one being the music on hold, ringing and silence. Why would axon play the music on hold before transfering to ivm? would this be a server issue (to slow) of a software issue that i have missed somewhere?

    MOH plays for a second before IVM OGM plays

    in IVM

    Posted

    I know ive seen this before.

     

    When a called calls axon picksup and transfer to MOH then to IVM. I have it set to only goto group 701 and has only one ext 198. it works fine on another system but when we changed over to a newer system this happened. I know when it was first set up it had this issue but cant find where to correct it.

     

    Thank

    ET BE and multi line call in

    in Express Talk

    Posted

    Yes, Express Talk can support up to 6 incoming phone calls at a time.

     

    However, the amount of incoming phone calls may be restricted by other factors (such as limits set by your VoIP provider). Often outbound calls are completely separate to inbound, and some providers will for example only give you 1 incoming channel/line but provide you with for example 3 outbound lines/channels. Many services are reluctant to give you several incoming channels especially if you're on a flat rate plan for the DID because they will get not benefits from allowing so many calls, in fact they may actually lose out from the costs of bandwidth and whatnot from those calls.

     

    Also, are you using Axon in between? Or are you connecting ET directly to the VoIP provider? Is the Voicemail system on your end (e.g. IVM)? Or is it going to the Voicemail system from your provider (this will help establish whether the provider is limiting the calls, or if the problem lies internally).

     

    I have axon 1.20 and et connects through axon, i do have ivm calls directed to ine ext and it transfers those calls. i understand about the line in use so thats not a big problem.

    grandstream 2000

    in Axon

    Posted

    Hi Dan

     

    I don't think the music on hold is configurable.

     

    I use Grandstream 2000s and music on hold works, but I don't know of a way to change it!

    can you post or send me a copy of your basic, advanced and account 1 setting so i can veiw the differances?

    SPA3000 / SPA3102 GUIDE

    in Axon

    Posted

    didnt have the issue with the not answering for the error you had but i created the 198 external line to ring on group 198 and it worked dont know why but it did. I hopes its not a bug and the next version its fixed and it breaks the current setup. well see in the next test inviroment. :rolleyes:

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