towns
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Posts posted by towns
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The default for the Voicemail duration field under the domain settings appears to be blank. How long of a message can someone leave if you keep this field blank?
Thanks for your help
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I am running 2.1.6.2450. When you call our hunt group the phone on the callers side dosen't ring. There is just silence until someone answers. Has anyone ran into this problem before? It is a DID to this hunt group. If I dial the main number and then dial that hunt group extension, the phone rings on the callers end.
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Hi towns,
Could you please the change following settings on your PBX ?
Under "Settings->General" page go to "SIP Settings" section. Select radio button "Long" for "Use Short SIP Headers" and hit "Save".
Now restart the PAC. Let me know how it goes.
Pradeep
That fixed it.
Thanks for the help,
Towns
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Does anyone have any other ideas?
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Okay, not beautiful, but it seems that you have an Extension name which is not numeric... Don't ask me why, but that is what is in the code! Can you pick a Extension that is a number? [*blush*]
I changed the pac extension from pac to 41. This seems to have gotten rid of the error but, I still don’t have anything in the extension console. This is the latest log file.
Message type :INFORMATION received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
Connected to 192.168.100.4
-----------------------End Message--------------------------
Message type :OUTBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
REGISTER sip:192.168.100.4 SIP/2.0
Via: SIP/2.0/TCP 192.168.100.126:4051
From:<sip:41@192.168.100.4>
To:<sip:41@192.168.100.4>
Call-ID: b7da7579-dded-472c-85bf-57b981430f6f
CSeq:15 REGISTER
Max-Forwards: 70
Contact: <sip:41@192.168.100.126:4051;transport=tcp;reg-id=1>;q=1.0;+sip.instance="<urn:uuid:dfbf5602-c710-484d-9624-7d0378a3a851>"
User-Agent: pbxnsip AC/0.1
Supported: buttons
Authorization: Digest username="41",realm="192.168.100.4",nonce="72bad5f7c9d5316896848f8b06e16d52",uri="sip:192.168.100.4",response="110af02ec763b2f71397063e331ec21f",algorithm=MD5
Expires:3600
Content-Length: 0
-----------------------End Message--------------------------
Message type :INBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
SIP/2.0 401 Authentication Requiredv: SIP/2.0/TCP 192.168.100.126:4051f: <sip:41@192.168.100.4>t: <sip:41@192.168.100.4>;tag=e8c8282c35i: b7da7579-dded-472c-85bf-57b981430f6fCSeq: 15 REGISTERUser-Agent: pbxnsip-PBX/2.2.1.2757WWW-Authenticate: Digest realm="192.168.100.4",nonce="52fcfc510d345d683995991f3c1fe608",domain="sip:192.168.100.4",algorithm=MD5l: 0
-----------------------End Message--------------------------
Message type :OUTBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
REGISTER sip:192.168.100.4 SIP/2.0
Via: SIP/2.0/TCP 192.168.100.126:4051
From:<sip:41@192.168.100.4>
To:<sip:41@192.168.100.4>
Call-ID: b7da7579-dded-472c-85bf-57b981430f6f
CSeq:16 REGISTER
Max-Forwards: 70
Contact: <sip:41@192.168.100.126:4051;transport=tcp;reg-id=1>;q=1.0;+sip.instance="<urn:uuid:a81e0ea5-b9e8-4d22-8c92-403b2c15f97f>"
User-Agent: pbxnsip AC/0.1
Supported: buttons
Authorization: Digest username="41",realm="192.168.100.4",nonce="52fcfc510d345d683995991f3c1fe608",uri="sip:192.168.100.4",response="66063f8732eaf04aa9f60603d4495b24",algorithm=MD5
Expires:3600
Content-Length: 0
-----------------------End Message--------------------------
Message type :INBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
SIP/2.0 200 Ok
v: SIP/2.0/TCP 192.168.100.126:4051
f: <sip:41@192.168.100.4>
t: <sip:41@192.168.100.4>;tag=e8c8282c35
i: b7da7579-dded-472c-85bf-57b981430f6f
CSeq: 16 REGISTER
m: <sip:41@192.168.100.126:4051;transport=tcp;reg-id=1>;expires=180
l: 0
-----------------------End Message--------------------------
Message type :INBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
MESSAGE sip:41@192.168.100.126:4051;transport=tcp;reg-id=1 SIP/2.0
v: SIP/2.0/TCP 192.168.100.4:5060;branch=z9hG4bK-cc28cc146db324985cb882a1b67ece75;rport
f: "attendant console" <sip:41@localhost>;tag=51695
t: "attendant console" <sip:41@localhost>
i: 0dv1b4fc@pbx
CSeq: 27957 MESSAGE
Max-Forwards: 70
m: <sip:192.168.100.4:5060;transport=tcp>
Subject: buttons
c: application/x-buttons
l: 247
-----------------------End Message--------------------------
Message type :INBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
k=1
x=ext
l=201
s=offline
d=Jack Smith
-----------------------End Message--------------------------
Message type :OUTBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
SIP/2.0 503 Not Implemented
Via: SIP/2.0/TCP 192.168.100.126:4051;branch=
Call-ID:
CSeq:16 INVITE
Contact: <sip:41@192.168.100.126:4051;transport=tcp;reg-id=1>;q=1.0;+sip.instance="<urn:uuid:2bf8afbf-ddfe-4321-8d55-0308c1ef9c01>"
User-Agent: pbxnsip AC/0.1
Content-Length: 0
-----------------------End Message--------------------------
Message type :INBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
k=2
x=ext
l=202
m=0/0
s=offline
d=John Smith
-----------------------End Message--------------------------
Message type :OUTBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
SIP/2.0 503 Not Implemented
Via: SIP/2.0/TCP 192.168.100.126:4051;branch=
Call-ID:
CSeq:16 INVITE
Contact: <sip:41@192.168.100.126:4051;transport=tcp;reg-id=1>;q=1.0;+sip.instance="<urn:uuid:be5ee5f8-dc93-4691-adaa-e23864ce4deb>"
User-Agent: pbxnsip AC/0.1
Content-Length: 0
-----------------------End Message--------------------------
Message type :INBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
k=3
x=ext
l=203
m=0/0
s=offline
d=Joe Smith
-----------------------End Message--------------------------
Message type :OUTBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
SIP/2.0 503 Not Implemented
Via: SIP/2.0/TCP 192.168.100.126:4051;branch=
Call-ID:
CSeq:16 INVITE
Contact: <sip:41@192.168.100.126:4051;transport=tcp;reg-id=1>;q=1.0;+sip.instance="<urn:uuid:b6748d7e-9192-46e6-bedd-bba4add07ced>"
User-Agent: pbxnsip AC/0.1
Content-Length: 0
-----------------------End Message--------------------------
Message type :INBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
k=4
x=ext
l=204
m=0/0
s=offline
d=Gary Smith
-----------------------End Message--------------------------
Message type :OUTBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
SIP/2.0 503 Not Implemented
Via: SIP/2.0/TCP 192.168.100.126:4051;branch=
Call-ID:
CSeq:16 INVITE
Contact: <sip:41@192.168.100.126:4051;transport=tcp;reg-id=1>;q=1.0;+sip.instance="<urn:uuid:3c2ae448-2748-439c-bdac-2cca5bdfb7c9>"
User-Agent: pbxnsip AC/0.1
Content-Length: 0
-----------------------End Message--------------------------
Message type :INBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
k=5
x=ext
l=205
m=0/2
d=Jerry Smith
-----------------------End Message--------------------------
Message type :OUTBOUND received at :4/30/2008 11:46:15 AM
-----------------------Begin Message------------------------
SIP/2.0 503 Not Implemented
Via: SIP/2.0/TCP 192.168.100.126:4051;branch=
Call-ID:
CSeq:16 INVITE
Contact: <sip:41@192.168.100.126:4051;transport=tcp;reg-id=1>;q=1.0;+sip.instance="<urn:uuid:5059048a-e313-4e0b-a492-0883891b124a>"
User-Agent: pbxnsip AC/0.1
Content-Length: 0
-----------------------End Message--------------------------
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Oh it seems you are using an old version of the PAC. This version had a bug with the parsing of the responses, and was not able to safely locate the attachments (sic!). Try this link: http://www.pbxnsip.com/pac/setup.exe, that should get you the latest and greatest.
I uninstalled the pac that I had on my pc. I then installed the version from the link in the previous reply. I still don't have anything showing up in my extension tab. Also, after I installed the pac the version is still showing up as 1.9.1.0. I am running 2.2.1.2757 (Win32) Version of pbxnsip
This is the error I get in the messages tab of the pac console. I don't get an error message at all if I don't assign the pac extension a configuration profile
Message type :ERROR received at :4/30/2008 10:39:22 AM
-----------------------Begin Message------------------------
System.FormatException: Input string was not in a correct format.
at System.Number.StringToNumber(String str, NumberStyles options, NumberBuffer& number, NumberFormatInfo info, Boolean parseDecimal)
at System.Number.ParseInt32(String s, NumberStyles style, NumberFormatInfo info)
at System.Int32.Parse(String s)
at AttendantConsole.MainForm.send501(String message)
-----------------------End Message--------------------------
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Did you set up a button group for the extension that your register with?
Yes, I tried a restart and, I have the PAC buttons setup. In the PAC extension, I chose the PAC configuration Profile.
I included a word doc of the pac log
FYI: Extension 205 is the only extension that has a phone registered to it.
This is my button list for the PAC
Button List "PAC"
name type parameter
1 Monitor Ext. (on software ext. board) 201
2 Monitor Ext. (on software ext. board) 202
3 Monitor Ext. (on software ext. board) 203
4 Monitor Ext. (on software ext. board) 204
5 Monitor Ext. (on software ext. board) 205
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I have installed the pac onto my pc. It is registering and under the information tab it shows all the extensions that I setup for the pac buttons. The problem I am having is it doesn't show any thing in the extention tab. Is their something else I need to setup? I will attach a screan shot of the pac extension tab. Also, the link I downloaded the pac from is supposed to be version 1.9.2.0 but, the version I have after install is 1.9.1.0. Thanks for any help
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We have a CS410 using the 4 FXO ports. Each FXO port has a different DID number. These DID numbers also match aliases on four accounts. We have discovered that when the main number is called by two different phones simultaneously, one call is answered by the auto-attendant (this is the correct behavior) but the second call goes right to an account that matches the DID number on the second FXO port (not the correct behavior). If three calls are placed simultaneously, then the same behavior occurs for calls one and two and the third call goes to the DID number on the third FXO port. This does not seem like the correct behavior but we have not been able to find a way to correct this. Have we missed something
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Maybe just put the xxxxxxxxxx there.
I tried that.
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Did you try playing with the Remote Party/Privacy Indication:on the trunk?
Yes I have tried all of the settings in the Remote Party/Privacy Indication
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Do you remember what you had in that field before? ...
Before we upgraded to the latest version, I had a$ xxxxxxxxxx in the Explicit Remote-Party-ID. Again the x's representing our actual phone number.
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We got rid of these "$f $i $x $y $z" strings in the trunk. I think the new name is now "Trunk DID". Normally, it is okay to just leave it empty, the PBX then will automatically fill something useful in.
It is debatable if that was a smart move. On the one hand, it caused a lot of support and problems with too much programmablility for nothing. On the other hand we have an upgrade issue, and now we need to explain everyone
.I tried to leave the Trunk DID blank, but the caller id when receiving a call from my voicemail came up as unknown again. So is there something I can do about this problem?
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I looked at the wiki a number of times for this problem. I will try to explain my problem with a little more detail. I was running pbxnsip version 2.0.3 and moved to a 2.1.xx. When I call directly from a phone snom 320 to my cell phone, the caller id shows the tel: alias for the extension I am dialing from. When I leave a message in my mailbox and have the message redirected to my cell phone the caller id shows up as unknown. In the earlier version 2.0.3, the call from my mailbox showed up as the tel: alias of my extension. If I have the call redirected directly to my cell phone and skip the mailbox I get the caller id of the original inbound call. What has changed in the new version? Am I just missing a setting? If I can give you more information please let me know. This is a screen shot of part of my trunk settings for out bound calls. Instead of xxxxxxxxxx for my trunk DID I have the main of our office.
FYI: If I have the call redirected directly to my cell phone and skip the mailbox I get the caller id of the original inbound call (which is fine for my).
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When my voicemail calls my cell phone letting me know I have a message, it shows up as unknown. How do I fix this so it will show the number as if I was just dialing from the office? It was working fine until I upgraded to the latest firmware.
Having the TEL: alias in each account was set and prior to moving up from 2442 up to 2449 the redirected call was correctly seeing the caller ID of a redirected call.
Did something change that affects the outbound caller ID presentation to our SIP provider?
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The RAM is not the problem here. Recordings are written to the file system. The 410 has 256 MB (minus a few MB for itself), which could keep a couple of hours.
You can also use a sip URI to offload the recording to an external SIP-compliant recording system.
If you want to do serious recording, IMHO you must use a PC. The embedded system is not good in storing large amounts of data and it is not easy to pull the data off the box.
I am looking for the settings in the web interface of the cs410 that will allow me to set what calls I would like to record. I have a windows version of this software and, I am able to navigate to the page that allows me to choose what type of calls I want to record. I am unable to find these options in the cs410 linux version. Am I just in the wrong spot or does this version not offer the same options for recording calls?
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Did you see http://wiki.pbxnsip.com/index.php/Recording?
I have a windows version of this software running version 2.1.6.2442 (win32) and under domain, settings I'm given this option:
Recording default for this domain:
Record incoming calls from hunt group: YesNo
Record incoming calls from agent group: YesNo
Record incoming calls from extension: YesNo
Record outgoing calls to internal numbers: YesNo
Record outgoing calls to external numbers: YesNo
I was able to record a call and find it in the directory.
I also have another pbx running version 2.1.6.2448 (linux) I cannot find the same options on this pbx.
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I am trying to record all incoming and outgoing calls to a certain extension. I am trouble figuring out how to do this. I also would like to know where the recorded calls are stored and how to have the recorded calls sent to a specific email address.
Thanks
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I am setting up a dial plan for a remote office. The home office has 3 digit extensions. If I use 5xxx the call won't go through if I use 5* the call does go through. Since the 5* dial plan goes through I'm pretty sure I have my trunk set up correctly. I don't have anything in the replacement box on 5xxx or 5*. I don't want to use the 5* because if I dial a 7 digit number that starts with 5 it will go out the wrong trunk. Does anyone know how to fix this problem?
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PAC update
in PAC and WAC
Posted
I updated to a new version of the PAC Version 1.9.2.12. I don't have my extensions tab anymore. The only tabs are treeview, settings, messages. Everything is still working fine. Is the extension view not supposed to be in this version?