mybusinessvoice
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Posts posted by mybusinessvoice
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can you please give me an example in point form or send me a link on how to set this up
thanks
Leigh
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hi
Is there a way to setup several modes on the system for example can i setup 3 modes 1. holiday 2. day 3. weekend forward the trunk to this mode then put a mode key on a BLF key so the customer can toggle between modes
this would be a great feature as i'm always getting calls from my customers to setup xmas, easter messages and diversions
Thanks
leigh
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Hi All,
Has someone connected a Epygi FXO gateway to a SoHo as i have tried and failed, i would like to connect it for incoming and outgoing calls
thanks
Leigh
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YES it works thank you, with it set to message 180 will this be a problem in the future for any other features?
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great can you tell me what i need to turn off because i cant find anything called SDP in the trunk settings
Thanks
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Can you please help me with this problem i still have incoming calls drop out after 3-4 rings
Logfile:
Clear or Reload the log.
[7] 2011/09/17 10:27:39: SIP Rx udp:203.176.185.10:5060:
CANCEL sip:61298993799@119.252.88.194:5060;transport=udp;line=182be0c5 SIP/2.0
Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bK4602.0865205332bb78e8c86dd7016a6df7c9.0
Via: SIP/2.0/UDP 203.176.185.10:5061;rport=5061;branch=z9hG4bK6385b86ed87d03ed097262b49db7069e
Max-Forwards: 16
From: <sip:0414352401@203.176.185.10>;tag=b64bc8384a85907ad21d3d20eb61e8e3
To: <sip:61298993799@203.176.185.10>
Call-ID: call-F1789020-34C3-2E10-0A0D-278CA@203.176.186.11~1o
CSeq: 200 CANCEL
Expires: 300
User-Agent: Sippy
[7] 2011/09/17 10:27:39: SIP Tx udp:203.176.185.10:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bK4602.0865205332bb78e8c86dd7016a6df7c9.0
Via: SIP/2.0/UDP 203.176.185.10:5061;rport=5061;branch=z9hG4bK6385b86ed87d03ed097262b49db7069e
From: <sip:0414352401@203.176.185.10>;tag=b64bc8384a85907ad21d3d20eb61e8e3
To: <sip:61298993799@203.176.185.10>;tag=bd9c12b372
Call-ID: call-F1789020-34C3-2E10-0A0D-278CA@203.176.186.11~1o
CSeq: 200 CANCEL
Contact: <sip:61298993799@119.252.88.194:5060;transport=udp>
User-Agent: Mybusinessvoice/4.0.1.3499
Content-Length: 0
[7] 2011/09/17 10:27:39: SIP Tx udp:203.176.185.10:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bK4602.0865205332bb78e8c86dd7016a6df7c9.0
Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK6385b86ed87d03ed097262b49db7069e;rport=5061
Record-Route: <sip:203.176.185.10;ftag=b64bc8384a85907ad21d3d20eb61e8e3;lr>
From: <sip:0414352401@203.176.185.10>;tag=b64bc8384a85907ad21d3d20eb61e8e3
To: <sip:61298993799@203.176.185.10>;tag=bd9c12b372
Call-ID: call-F1789020-34C3-2E10-0A0D-278CA@203.176.186.11~1o
CSeq: 200 INVITE
Contact: <sip:61298993799@119.252.88.194:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: Mybusinessvoice/4.0.1.3499
Content-Length: 0
[7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:2048:
CANCEL sip:100@165.228.88.83:2048;line=uxvspv5f SIP/2.0
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-b6f5d0a576a3752b8b3510b6cebfcc49;rport
From: <sip:0414352401@203.176.185.10;user=phone>;tag=18577
To: <sip:61298993799@203.176.185.10;user=phone>
Call-ID: 95295cf1@pbx
CSeq: 4223 CANCEL
Max-Forwards: 70
Content-Length: 0
[7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:1024:
CANCEL sip:101@165.228.88.83:1024;line=cozyryzk SIP/2.0
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fbaa22b07c9e5fcda08b0e1dcae35b45;rport
From: <sip:0414352401@203.176.185.10;user=phone>;tag=49205
To: <sip:61298993799@203.176.185.10;user=phone>
Call-ID: 60769716@pbx
CSeq: 10 CANCEL
Max-Forwards: 70
Content-Length: 0
[7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:1026:
CANCEL sip:102@165.228.88.83:1026;line=ybc60i8q SIP/2.0
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-070d3ac9608ccb9742d1d9e49b19c814;rport
From: <sip:0414352401@203.176.185.10;user=phone>;tag=62968
To: <sip:61298993799@203.176.185.10;user=phone>
Call-ID: 9885769c@pbx
CSeq: 23188 CANCEL
Max-Forwards: 70
Content-Length: 0
[7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:1025:
CANCEL sip:103@165.228.88.83:1025;line=bpetirz2 SIP/2.0
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fcd6a3e79831ca2c4991cc4f09818cb7;rport
From: <sip:0414352401@203.176.185.10;user=phone>;tag=7414
To: <sip:61298993799@203.176.185.10;user=phone>
Call-ID: 0262a24e@pbx
CSeq: 16007 CANCEL
Max-Forwards: 70
Content-Length: 0
[7] 2011/09/17 10:27:39: SIP Rx udp:203.176.185.10:5060:
ACK sip:61298993799@119.252.88.194:5060;transport=udp;line=182be0c5 SIP/2.0
Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bK4602.0865205332bb78e8c86dd7016a6df7c9.0
From: <sip:0414352401@203.176.185.10>;tag=b64bc8384a85907ad21d3d20eb61e8e3
Call-ID: call-F1789020-34C3-2E10-0A0D-278CA@203.176.186.11~1o
To: <sip:61298993799@203.176.185.10>;tag=bd9c12b372
CSeq: 200 ACK
User-Agent: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
[8] 2011/09/17 10:27:39: Hangup: Call call-F1789020-34C3-2E10-0A0D-278CA@203.176.186.11~1o#bd9c12b372 not found
[7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:2048:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-b6f5d0a576a3752b8b3510b6cebfcc49;rport=5060
From: <sip:0414352401@203.176.185.10;user=phone>;tag=18577
To: <sip:61298993799@203.176.185.10;user=phone>;tag=0r2e4lbqfr
Call-ID: 95295cf1@pbx
CSeq: 4223 CANCEL
Content-Length: 0
[7] 2011/09/17 10:27:39: Call 95295cf1@pbx#18577: Clear last request
[7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:1024:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fbaa22b07c9e5fcda08b0e1dcae35b45;rport=5060
From: <sip:0414352401@203.176.185.10;user=phone>;tag=49205
To: <sip:61298993799@203.176.185.10;user=phone>;tag=l75r3t8le3
Call-ID: 60769716@pbx
CSeq: 10 CANCEL
Content-Length: 0
[7] 2011/09/17 10:27:39: Call 60769716@pbx#49205: Clear last request
[7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:1026:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-070d3ac9608ccb9742d1d9e49b19c814;rport=5060
From: <sip:0414352401@203.176.185.10;user=phone>;tag=62968
To: <sip:61298993799@203.176.185.10;user=phone>;tag=lvaq3si9p0
Call-ID: 9885769c@pbx
CSeq: 23188 CANCEL
Content-Length: 0
[7] 2011/09/17 10:27:39: Call 9885769c@pbx#62968: Clear last request
[7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:2048:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-b6f5d0a576a3752b8b3510b6cebfcc49;rport=5060
From: <sip:0414352401@203.176.185.10;user=phone>;tag=18577
To: <sip:61298993799@203.176.185.10;user=phone>;tag=0r2e4lbqfr
Call-ID: 95295cf1@pbx
CSeq: 4223 INVITE
Contact: <sip:100@165.228.88.83:2048;line=uxvspv5f>;reg-id=1
Content-Length: 0
[7] 2011/09/17 10:27:39: Call 95295cf1@pbx#18577: Clear last INVITE
[7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:2048:
ACK sip:100@165.228.88.83:2048;line=uxvspv5f SIP/2.0
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-b6f5d0a576a3752b8b3510b6cebfcc49;rport
From: <sip:0414352401@203.176.185.10;user=phone>;tag=18577
To: <sip:61298993799@203.176.185.10;user=phone>;tag=0r2e4lbqfr
Call-ID: 95295cf1@pbx
CSeq: 4223 ACK
Max-Forwards: 70
Contact: <sip:100@119.252.88.194:5060;transport=udp>
Content-Length: 0
[5] 2011/09/17 10:27:39: INVITE Response 487 Request Terminated: Terminate 95295cf1@pbx
[7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:1025:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fcd6a3e79831ca2c4991cc4f09818cb7;rport=5060
From: <sip:0414352401@203.176.185.10;user=phone>;tag=7414
To: <sip:61298993799@203.176.185.10;user=phone>;tag=aob35ydmc0
Call-ID: 0262a24e@pbx
CSeq: 16007 CANCEL
Content-Length: 0
[7] 2011/09/17 10:27:39: Call 0262a24e@pbx#7414: Clear last request
[7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:1024:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fbaa22b07c9e5fcda08b0e1dcae35b45;rport=5060
From: <sip:0414352401@203.176.185.10;user=phone>;tag=49205
To: <sip:61298993799@203.176.185.10;user=phone>;tag=l75r3t8le3
Call-ID: 60769716@pbx
CSeq: 10 INVITE
Contact: <sip:101@165.228.88.83:1024;line=cozyryzk>;reg-id=1
Content-Length: 0
[7] 2011/09/17 10:27:39: Call 60769716@pbx#49205: Clear last INVITE
[7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:1024:
ACK sip:101@165.228.88.83:1024;line=cozyryzk SIP/2.0
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fbaa22b07c9e5fcda08b0e1dcae35b45;rport
From: <sip:0414352401@203.176.185.10;user=phone>;tag=49205
To: <sip:61298993799@203.176.185.10;user=phone>;tag=l75r3t8le3
Call-ID: 60769716@pbx
CSeq: 10 ACK
Max-Forwards: 70
Contact: <sip:101@119.252.88.194:5060;transport=udp>
Content-Length: 0
[5] 2011/09/17 10:27:39: INVITE Response 487 Request Terminated: Terminate 60769716@pbx
[7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:1026:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-070d3ac9608ccb9742d1d9e49b19c814;rport=5060
From: <sip:0414352401@203.176.185.10;user=phone>;tag=62968
To: <sip:61298993799@203.176.185.10;user=phone>;tag=lvaq3si9p0
Call-ID: 9885769c@pbx
CSeq: 23188 INVITE
Contact: <sip:102@165.228.88.83:1026;line=ybc60i8q>;reg-id=1
Content-Length: 0
[7] 2011/09/17 10:27:39: Call 9885769c@pbx#62968: Clear last INVITE
[7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:1026:
ACK sip:102@165.228.88.83:1026;line=ybc60i8q SIP/2.0
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-070d3ac9608ccb9742d1d9e49b19c814;rport
From: <sip:0414352401@203.176.185.10;user=phone>;tag=62968
To: <sip:61298993799@203.176.185.10;user=phone>;tag=lvaq3si9p0
Call-ID: 9885769c@pbx
CSeq: 23188 ACK
Max-Forwards: 70
Contact: <sip:102@119.252.88.194:5060;transport=udp>
Content-Length: 0
[5] 2011/09/17 10:27:39: INVITE Response 487 Request Terminated: Terminate 9885769c@pbx
[7] 2011/09/17 10:27:39: SIP Rx udp:165.228.88.83:1025:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fcd6a3e79831ca2c4991cc4f09818cb7;rport=5060
From: <sip:0414352401@203.176.185.10;user=phone>;tag=7414
To: <sip:61298993799@203.176.185.10;user=phone>;tag=aob35ydmc0
Call-ID: 0262a24e@pbx
CSeq: 16007 INVITE
Contact: <sip:103@165.228.88.83:1025;line=bpetirz2>;reg-id=1
Content-Length: 0
[7] 2011/09/17 10:27:39: Call 0262a24e@pbx#7414: Clear last INVITE
[7] 2011/09/17 10:27:39: SIP Tx udp:165.228.88.83:1025:
ACK sip:103@165.228.88.83:1025;line=bpetirz2 SIP/2.0
Via: SIP/2.0/UDP 119.252.88.194:5060;branch=z9hG4bK-fcd6a3e79831ca2c4991cc4f09818cb7;rport
From: <sip:0414352401@203.176.185.10;user=phone>;tag=7414
To: <sip:61298993799@203.176.185.10;user=phone>;tag=aob35ydmc0
Call-ID: 0262a24e@pbx
CSeq: 16007 ACK
Max-Forwards: 70
Contact: <sip:103@119.252.88.194:5060;transport=udp>
Content-Length: 0
[5] 2011/09/17 10:27:39: INVITE Response 487 Request Terminated: Terminate 0262a24e@pbx
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Hi
I have an old CS410 and i have forgot the password how do i default the system so i can login with the default username and password?
Thanks
Leigh
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Ok thanks, my system is a hosted PBX with multiple domains as the system grows in extensions do i need to increase the global settings in the system to handle more traffic? if so which ones. Because this problem has only just started as ive increase the number of domains and extensions
Thanks
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im running firmware 4.0.1.3499 is this a problem?
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i have a hosted rental lic with snom plus i tested the system when there was know one on it
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Hi,
I have a hosted PBX, all of my domains are having the same problem with calls coming into the PBX cut off after about 3-4 rings but 50% of the time the call can ring and go to voicemail
Sip logs
[5] 2011/09/14 21:50:37: Domain trunk Second trunk@sydneydentalprofessionals sends call to 190 in domain sydneydentalprofessionals
[7] 2011/09/14 21:50:37: Call 2403e639@pbx#2049: Clear last request
[7] 2011/09/14 21:50:37: Call 0e3adfe2@pbx#9137: Clear last request
[8] 2011/09/14 21:50:47: Hangup: Call call-F1835DA7-4AC1-2E10-1D0C-4EF93@203.176.186.12~1o#7189fba86e not found
[7] 2011/09/14 21:50:47: Call 2403e639@pbx#2049: Clear last request
[7] 2011/09/14 21:50:47: Call 0e3adfe2@pbx#9137: Clear last request
[7] 2011/09/14 21:50:47: Call 2403e639@pbx#2049: Clear last INVITE
[5] 2011/09/14 21:50:47: INVITE Response 487 Request Terminated: Terminate 2403e639@pbx
[7] 2011/09/14 21:50:47: Call 0e3adfe2@pbx#9137: Clear last INVITE
[5] 2011/09/14 21:50:47: INVITE Response 487 Request Terminated: Terminate 0e3adfe2@pbx
All my extensions and trunk are set to G729
Thanks
Leigh
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Hi,
I have a hosted PBX setup,
Is there a way to send each recording to an email address, if not how has my customer get access to their recordings.
Thanks
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hi,
I cant get the Bria softphone app to register to my snomone PBX can you please explain how to register it step by step
Thanks
leigh
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Hi,
A strange thing is happening with the system i have auto provisioned all the extension working fine. As normal practice i configure each extension with the customers name.
Example
101 chris
102 steven
103 john
104 harry
105 joe
now the problem is no matter what phone i dial from it displays on the snom phone that im calling joe when im calling chris, the phone displays chris for like .5sec then displays joe.
the snom phone are 821 using fireware 8.4.31
Thanks
leigh
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Works great thanks
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Hi,
I have auto provisioned my snom handsets to the Snomone PBX, programmed the PBX for Key system configuration. i have set co1 co2 co3 into the trunk then i made a button profile with button 1 to co1 2 to co2 3 to co3
Then i asigned the buttons to each extension, line 1 on hold i can pick up line 1 from any other phone. Great!
But when i put a call on hold from any phone i get the hold tone through the speaker only, the only way to get rid of the hold tone from the speaker is to take the call off hold.
The way it should work is once the call is placed on hold the phone just sit their quite not making a loud hold tone.
Thanks
Leigh
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I have turned off loop detection and restarted the PBX but its still disconnecting the call
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Where do i find the 'max_loop' global parameter on the PBX
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Hi
How do i set up remote access to an internal extension so the customer can change his voice mail message from his cell phone
Thanks
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Hi,
I'm having trouble with what i think should be a very simple setup
i have setup the following, a trunk comes into the system which hits a hunt group Ext 440 which is set up as-
Stages:
Stage 1 Extensions:101 102 Duration 10 seconds
Stage 2 Extensions: Duration
Stage 3 Extensions: Duration
Final Stage:400(which is a auto attendant)
the auto attendant gives the caller 2 options 1 (8101) to leave a message and 2 which forwards to a second hunt group.
Extension 441
Stages:
Stage 1 Extensions:101 102 Duration 10 seconds
Stage 2 Extensions: Duration
Stage 3 Extensions: Duration
Final Stage:8101 When the call reaches the final stage it disconnects the call it should go to extension 101 voicemail box
please help
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Hi
I have my system setup as a hosted system and im trying to setup auto PNP with the snom redirction server for touch free config
snom has set this up at their end and gave me a new key, so what do i need to do to make it work?
I tryed binding the MAC of the phone then resetting the phone but no luck.
Thanks
Leigh
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Hi All,
Can some please explain how i setup auto plug and play, i have PBXNSIP setup as a hosted system. All my handsets are remote on the customers site.
so can i program the system so when i plug the snom phone in for the first time it redirect's to my system and get its configuration without me touching the phone.
thanks
Leigh
-
Hi
Can you please send me the link to download snom one admin guide
Thanks
-
I have a customer that had an old NEC key system with PSTN lines i have connected them to my PBXNSIP hosted system v4 with snom 320. My customer is asking for key mode simulation e.g. co lines that appear and all handsets if i take a call and put that call on hold all phone will see that call on hold just like the old NEC key system.
I have followed instructions from http://kiwi.pbxnsip.com/index.php/Shared_Line_Appearances with no luck
Thank you
Password lost
in Embedded
Posted
how do i login via SSH?