CarlH
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Posts posted by CarlH
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This didn't work for me. I made two accounts 43 0842013043 on domain01 and 30 0842016130 on domain02
I still cant call cross domains wit the alias. Do I need to cahnge anything in dialpans or trunks?
I finally got it working! I used both alias on each extension and "try loopback" in the dialplan. I can now call the alias I set in other domains.
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This didn't work for me. I made two accounts 43 0842013043 on domain01 and 30 0842016130 on domain02
I still cant call cross domains wit the alias. Do I need to cahnge anything in dialpans or trunks?
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We previously used tel:alias to call numbers in other domains but since upgrading to version 3 this doesn't work anymore.
I read about the try loopback option as trunk but did not get it to work.
What i did was that I disabled loopback detection and then added "Try loopback" to the dialplan with the lowest "pref" and * as "pattern".
I'm guessing that's not all there is to it but since I can't find any more info about this I'm asking here
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Hi,
I would like some more info on this. Is it possible for you to write some type of step by step guide on how to configure cross domain calling? I have read several topics about this but haven't seen any clear information on what to do. Is it possible that you could write some kind of step by step because I don't think I'm the only one with this request.
Lets say we have two domains
domain1.net and domain2.net
in domain 1 we have nr 10-50
in domain 2 we have nr 10-50 as well
In the dial plan we currently have pattern * for the trunk.
Would it be possible to configure the dial plans so that you call domain2 by pressing eg #1 and then the nr eg #110 for nr 10 in domain2
One of the domains is localhost with domain01.net as alias name.
Br
Carl
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Edit: I'm posting this as a new topic under dialplans instead
Hi,
I would like som more info on this. Is it possible for you to write some type of step by step guide on how to configure inter domain calling?
Lets say we have two domains
domain1 and domain2.
in domain 1 we have nr 10-50
in domain 2 we have nr 10-50 aswell
In the dialplan we currently have pattern * for the trunk.
Would it be possible to configure the dialplans so that you call domain2 by pressing eg #1 and then the nr eg #110 for nr 10 in domain2
Br
Carl
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I would just take 0 (ulaw).
It works now
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Try forcing a specific codec on the trunk. Probably the provider has a problem when the PBX answeres with more than one codec (see http://wiki.pbxnsip.com/index.php/One-way_Audio).
Which codecs do you recommend?
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Looking at the old messages I must admit I don't exactly get what the problem is. Is it a problem related to RTP or is it a problem related to the phone number (+47 or 0047 or 01147 and so on)? Maybe you can get a fresh LOG...
Well, Ill try to explain more in detail. Our trunk provider has registered numbers in different countries in europe which are forwarded to our support line. All numbers except those for UK, Norway and Finland work fine. The provider has tried forwarding directly to one of their phones and it works. When those numbers are forwarded to our support line (an IVR) all we here is silence. The call is connected and I can see it in "calls" but we can't hear the IVR.
Denmark works Finland doesn't.
DENMARK (45) NATIONAL 1 +4569918175
FINLAND (358) HELSINKI (9) 1 +358942419025
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Yes, that whole topic is addressed in 3.1.1. If you can, get a 3-minute demo key, set up a test server and try the latest &
greatest (http://pbxnsip.com/protect/pbxctrl-3.1.1.3100.exe). Then set your contry code to 47 (if I am right here) and then the numbers should be formatting correctly - automatically.
Hi!
I have now upgraded to the latest version but we still have the same problem with numbers forwarded from Norway and UK.
Any thoughts? All help is very appreciated.
Br
Carl
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Which firmware version is the M3, I think the older fimware had a DTMF bug.
Firmware-Version: snom-m3-SIP/01.16//03-Jul-08 13:43
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Hi,
We recently bought a Snom M3 but we cant get DTMF to work with it. Any clue on what setting we should change?
Br
Carl
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Yeah, I know and i'm fully aware of that it was an extremely stupid thing to do. I too have done this procedure many times and never had any issues and of course the one time i didn't make a copy this happened
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I decided to try the latest version tonight after office hours in our live environment. I should NEVER have done that! All settings were deleted and the xml files were overwritten with 0 byte files.
I have never experienced this after an upgrade before. Luckily I had a copy thats a few weeks old of the PBX directory. I lost some extensions but I'm working now to restore them.
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Hi,
We are having problems with some international numbers that we have registered to forward calls to our helpdesk. E.g here are extracts from the log when calling to the Norwegian nr +4721031332 and to the Switzerland nr +41435000151. Both these nr are forwarded to 0842014000 which is the Swedish nr. When calling from the Norweigan nr i get connected but I don't hear anything. When I call to the Switz nr i hear the IVR loud and clear.
Any help would be greatly appreciated!
Here is a an extract from the log when calling to the Norwegian nr.
[7] 2008/12/02 09:24:03:SIP Rx udp:195.149.148.40:5060: BYE sip:0842014000@83.145.6.141:5060;transport=udp SIP/2.0
Record-Route: <sip:195.149.148.40;ftag=as7d8c22a2;lr=on>
Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bKa34d.10e4dbe6.0
Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK35364621;rport=5060
From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as7d8c22a2
To: <sip:4721031332@x.rtcfactory.com>;tag=f5150fa6b6
Call-ID: 11b0347b27f2a7164edf7e932101625f@195.138.212.41
CSeq: 103 BYE
User-Agent: RTC Gateway 2.0
Max-Forwards: 70
Content-Length: 0
P-hint: call from pstn gateway
[9] 2008/12/02 09:24:03:Resolve 29794699: aaaa udp 195.149.148.40 5060 [9] 2008/12/02 09:24:03:Resolve 29794699: a udp 195.149.148.40 5060 [9] 2008/12/02 09:24:03:Resolve 29794699: udp 195.149.148.40 5060 [7] 2008/12/02 09:24:03:SIP Tx udp:195.149.148.40:5060: SIP/2.0 200 Ok
Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bKa34d.10e4dbe6.0
Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK35364621;rport=5060
Record-Route: <sip:195.149.148.40;ftag=as7d8c22a2;lr=on>
From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as7d8c22a2
To: <sip:4721031332@x.rtcfactory.com>;tag=f5150fa6b6
Call-ID: 11b0347b27f2a7164edf7e932101625f@195.138.212.41
CSeq: 103 BYE
Contact: <sip:0842014000@83.145.6.141:5060;transport=udp>
User-Agent: pbxnsip-PBX/3.0.1.3023
RTP-RxStat: Dur=9,Pkt=430,Oct=73960,Underun=0
RTP-TxStat: Dur=9,Pkt=441,Oct=75852
Content-Length: 0
Here is a an extract from the log when calling to the Switz nr.
[7] 2008/12/02 09:10:59:SIP Rx udp:195.149.148.40:5060: INVITE sip:0842014000@83.145.6.141:5060;transport=udp;line=02e74f10 SIP/2.0
Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on>
Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bK6798.c5101c55.0
Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK72b6d9ec;rport=5060
From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486
To: <sip:41435000151@x.rtcfactory.com>
Contact: <sip:0046707960416@195.138.212.41>
Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41
CSeq: 102 INVITE
User-Agent: RTC Gateway 2.0
Max-Forwards: 70
Date: Tue, 02 Dec 2008 08:10:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 268
P-hint: call from pstn gateway
P-hint: local sip call
v=0
o=root 14929 14929 IN IP4 195.138.212.41
s=session
c=IN IP4 195.138.212.41
t=0 0
m=audio 15002 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[9] 2008/12/02 09:10:59:UDP: Opening socket on port 50050 [9] 2008/12/02 09:10:59:UDP: Opening socket on port 50051 [5] 2008/12/02 09:10:59:Identify trunk (line match) 27 [9] 2008/12/02 09:10:59:Resolve 29788667: aaaa udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788667: a udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788667: udp 195.149.148.40 5060 [7] 2008/12/02 09:10:59:SIP Tx udp:195.149.148.40:5060: SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bK6798.c5101c55.0
Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK72b6d9ec;rport=5060
Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on>
From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486
To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470
Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41
CSeq: 102 INVITE
Content-Length: 0
[7] 2008/12/02 09:10:59:Set packet length to 20 [6] 2008/12/02 09:10:59:Sending RTP for 033369853e67a81277f6e86855db9b4f@195.138.212.41#670aff6470 to 195.138.212.41:15002 [5] 2008/12/02 09:10:59:Trunk RTC 0842014000 sends call to 00 in domain smarthost.se [8] 2008/12/02 09:10:59:Play recordings/ivr79.wav [7] 2008/12/02 09:10:59:Set packet length to 20 [9] 2008/12/02 09:10:59:Resolve 29788668: aaaa udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788668: a udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788668: udp 195.149.148.40 5060 [7] 2008/12/02 09:10:59:SIP Tx udp:195.149.148.40:5060: SIP/2.0 200 Ok
Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bK6798.c5101c55.0
Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK72b6d9ec;rport=5060
Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on>
From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486
To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470
Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41
CSeq: 102 INVITE
Contact: <sip:0842014000@83.145.6.141:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.0.1.3023
Content-Type: application/sdp
Content-Length: 228
v=0
o=- 46299 46299 IN IP4 83.145.6.141
s=-
c=IN IP4 83.145.6.141
t=0 0
m=audio 50050 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[9] 2008/12/02 09:10:59:Resolve 29788669: aaaa udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788669: a udp 195.149.148.40 5060 [9] 2008/12/02 09:10:59:Resolve 29788669: udp 195.149.148.40 5060 [7] 2008/12/02 09:10:59:SIP Tx udp:195.149.148.40:5060: SIP/2.0 200 Ok
Via: SIP/2.0/UDP 195.149.148.40;branch=z9hG4bK6798.c5101c55.0
Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK72b6d9ec;rport=5060
Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on>
From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486
To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470
Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41
CSeq: 102 INVITE
Contact: <sip:0842014000@83.145.6.141:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.0.1.3023
Content-Type: application/sdp
Content-Length: 228
v=0
o=- 46299 46299 IN IP4 83.145.6.141
s=-
c=IN IP4 83.145.6.141
t=0 0
m=audio 50050 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[7] 2008/12/02 09:10:59:SIP Rx udp:195.149.148.40:5060: ACK sip:0842014000@83.145.6.141:5060;transport=udp SIP/2.0
Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on>
Via: SIP/2.0/UDP 195.149.148.40;branch=0
Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK5faf7a1a;rport=5060
From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486
To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470
Contact: <sip:0046707960416@195.138.212.41>
Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41
CSeq: 102 ACK
User-Agent: RTC Gateway 2.0
Max-Forwards: 70
Content-Length: 0
P-hint: call from pstn gateway
[7] 2008/12/02 09:10:59:SIP Rx udp:195.149.148.40:5060: ACK sip:0842014000@83.145.6.141:5060;transport=udp SIP/2.0
Record-Route: <sip:195.149.148.40;ftag=as6ebc1486;lr=on>
Via: SIP/2.0/UDP 195.149.148.40;branch=0
Via: SIP/2.0/UDP 195.138.212.41:5060;branch=z9hG4bK3cda2bc1;rport=5060
From: "0046707960416" <sip:0046707960416@195.138.212.41>;tag=as6ebc1486
To: <sip:41435000151@x.rtcfactory.com>;tag=670aff6470
Contact: <sip:0046707960416@195.138.212.41>
Call-ID: 033369853e67a81277f6e86855db9b4f@195.138.212.41
CSeq: 102 ACK
User-Agent: RTC Gateway 2.0
Max-Forwards: 70
Content-Length: 0
P-hint: call from pstn gateway
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The IP the server provisions is 10.10.120.2. Id like it to provision the dns name instead. The sonicwall PRO 3060 forwards necessary ports to 10.10.120.2
IPv4 Route Table =========================================================================== Interface List 0x1 ........................... MS TCP Loopback interface 0x10003 ...00 0d 60 9d 41 06 ...... Broadcom NetXtreme Gigabit Ethernet #2 0x10004 ...00 0d 60 9d 41 07 ...... Broadcom NetXtreme Gigabit Ethernet =========================================================================== =========================================================================== Active Routes: Network Destination Netmask Gateway Interface Metric 0.0.0.0 0.0.0.0 10.10.120.254 10.10.120.2 10 10.10.120.0 255.255.255.0 10.10.120.2 10.10.120.2 10 10.10.120.2 255.255.255.255 127.0.0.1 127.0.0.1 10 10.230.0.0 255.255.0.0 10.230.0.12 10.230.0.12 10 10.230.0.12 255.255.255.255 127.0.0.1 127.0.0.1 10 10.255.255.255 255.255.255.255 10.10.120.2 10.10.120.2 10 10.255.255.255 255.255.255.255 10.230.0.12 10.230.0.12 10 127.0.0.0 255.0.0.0 127.0.0.1 127.0.0.1 1 224.0.0.0 240.0.0.0 10.10.120.2 10.10.120.2 10 224.0.0.0 240.0.0.0 10.230.0.12 10.230.0.12 10 255.255.255.255 255.255.255.255 10.10.120.2 10.10.120.2 1 255.255.255.255 255.255.255.255 10.230.0.12 10.230.0.12 1 Default Gateway: 10.10.120.254 =========================================================================== Persistent Routes: None ipconfig /all Windows IP Configuration Host Name . . . . . . . . . . . . : Blade-01-02 Primary Dns Suffix . . . . . . . : xxxxxx.se Node Type . . . . . . . . . . . . : Unknown IP Routing Enabled. . . . . . . . : No WINS Proxy Enabled. . . . . . . . : No DNS Suffix Search List. . . . . . : xxxxxx.se Ethernet adapter Local Area Connection: Connection-specific DNS Suffix . : Description . . . . . . . . . . . : Broadcom NetXtreme Gigabit Ethernet #2 Physical Address. . . . . . . . . : 00-0D-60-9D-41-06 DHCP Enabled. . . . . . . . . . . : No IP Address. . . . . . . . . . . . : 10.10.120.2 Subnet Mask . . . . . . . . . . . : 255.255.255.0 Default Gateway . . . . . . . . . : 10.10.120.254 Ethernet adapter Local Area Connection 2: Connection-specific DNS Suffix . : Description . . . . . . . . . . . : Broadcom NetXtreme Gigabit Ethernet Physical Address. . . . . . . . . : 00-0D-60-9D-41-07 DHCP Enabled. . . . . . . . . . . : No IP Address. . . . . . . . . . . . : 10.230.0.12 Subnet Mask . . . . . . . . . . . : 255.255.0.0 Default Gateway . . . . . . . . . : DNS Servers . . . . . . . . . . . : 10.230.0.1 10.0.0.2 -
Hi,
We moved our pbxnsip server to a new server in new datacenter this weekend. Now when we try to provision our Polycom phones we get a server adress which we cant connect to. We want it to provision the phones with the public ip. Now it provisions phones with the private IP. How do we change that?
/Carl
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Hi,
After upgrading to 3.0 the outgoing numbers are wrong. When i call from extension 45 it shows +45 as caller when it should state the real number 0842013045. How do I correct this?
I have attached a picture of the trunk settings for 0842013045
Br
Carl
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Yea, sorry typo.
I know that for example AudioCodes, Vegastream, Mediatrix and also newer versions of Grandstream work with T.38. I think also Cisco Gateways support it. If you are looking for a sofphone that supports FAX (!) look at Zoiper. Did I forget a vendor?
We buy sip trunks from a VOIP provider and use them with PBXnSIP end then sell SIP accounts to our customers. We dont have any kind of appliance ourselves. All I need is some kind of t.38 converter for one fax machine. Do you know specific product that does this? Zoiper looks very interesting and we will defeniteley buy some of those. However this particular customer also needs to send fax.
BR
Carl
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Fro FAX with T.38, both the PSTN gateway and the FXS gateway needs to support it. If you are using a ITSP, then I would suggest to ask them if their switch supports T.38.
As for Fax to email, there is a softphone out there (name?) what supports T.38 and fax-to-email. Maybe someone remembers what the name was.
Could you be more specifik please? I didn't get any hits when googling Fro Fax.
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Hi!
I have customer who needs a fax solution. Today they have an ordinary FAX without T.38. Is it possible to use this fax with PBXnSIP with some kind of signal converter? If so I would greatly appreciate product recommendations for a small office.
I also wonder if its possible to implement some kind of FAX to PDF converter wich e-mails the fax as an pdf to the customer. Ive seen other VOIP providers providing this service.
Br
Carl
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Hi,
I can't find the TAPI add-on from the download page. I'm using the latest 2.1 RC version.
Thanks in advance!
Br
Carl
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Yea, we just changed that. Changing the To-header was technically a brilliant idea, however almost no phone does show the To-header. Now we changed it to the From-header and that should be in the 2.1 release candidate (hopefully coming out today).
Works perfectly with the new RC1. Thanks!
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Hi,
I'm having problems getting the groupname to show up on the phones when someone is calling one of our hunt groups. I have chosen Groupname with called number in the "To-header" option but it still doesn't show.
I'm guessing there's a setting somwhere that should solve this problem.
Thanks in avance.
Br
Carl
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Setting it to Remote-Party-ID solved the problem. Thanks!
. Thanks!
Dial plan for international calls
in Dial Plan Setup
Posted
H!
We have a customer in Sweden who are setting up a office in the UK. They want to use the pbxnsip in the UK office as well.
Today we have one trunk per account and the dial plan pattern is simply a *
Now we need to get some trunks in the UK and make sure that when the users dial Sweden +46 they will use the Swedish trunk otherwise the UK trunk. How do I accomplish this?
Regards
Carl