Vodia PBX
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Hi I have 1 trunk, with 4 ddi numbers on it. We currently have 3 extensions.
I want to make only one extesion ring when a specific ddi is called, and also i want that extension to dial out on that ddi.
Eg..
DII = 01611112222
Extension = 800
When I call 01611112222 i want only extension 800 to ring.
When i make a call from extension 800 i want it call out via 01611112222
The easiest way to deal with this is to use alias names that match the number. For example the extension would have the alias name 01611112222. Then on the trunk the setting "Send call to extension" can just stay empty. For outbound calls, set the ANI in the extension to what should be presented to the outside world.
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Sorry, do you mean the trunk configuration in pbxnsip? If so I've set-up outbound proxy to be ocs2007-b.i3q.local (our OCS mediation server). In OCS Mediation server the next hop PSTN is specified as the IP address of the pbxnsip box 10.150.100.207 on port 5060.
Yes, the trunk config in pbxnsip. That message above says "I got a request and I have no idea where it comes from". If the PBX cannot identify the trunk then there may be something wrong with the trunk outbound proxy. For example a famous problem is that the outbound proxy is set to the IP address of the host; when the request comes in the IP address of the request is 127.0.0.1, which does not match that IP address.
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... no, I don't see "received dtmf f", i get only:
[9] 20081021125333: DTMF: Power: 0 0 2 0 2 0 0 1 0
[9] 20081021125333: DTMF: Power: 1 2 0 0 0 2 0 0 1
[9] 20081021125333: DTMF: Power: 0 0 0 0 0 0 1 1 1
[9] 20081021125333: DTMF: Power: 0 2 0 0 0 2 0 0 1
.........
but nothing marked as "[6] received".......
I try also a call between 2 extensions (2 sip phone, snom and grandstream), once the call is established, I send a DTMF (only to try...), in the log i get the same messages, nothing like [6]...., only [9].....
Okay, at least we now know that the problem is the inband fax tone detection. Hmm. Maybe you can get a Wireshark trace, send me a private mail with a link where we can download it and then find out what the problem is. Lets see if we can detect the tone...
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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
Okay, in that case you need to take a look at http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses. You can override "IP Routing List". Replacing it with a DNS name sounds interesting, it might even work. Though then all user agents must be able to resolve DNS names, even for SIP Via headers. I would put a IP address there... If your IP address change, you can also easily change it in the replacement list.
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I create an IVR node with DTMF match list = !F!201! !T!250!
When a call arrives, if is a FAX, go to 201, else go to 250.
If I send a fax, the rule doesn't work, the call always go to 250.
What's wrong? There is something that I need to know?
Hmm. The pattern looks right to me. Do you see something like "[6] Received DTMF F" in the log file?
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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?
What does "route print" say? Anything surprising? The PBX generally uses a system call to find out what IP address it should present. Maybe the default IP gateway is a private IP address.
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[8] 2008/10/19 20:30:02: Could not find a trunk (2 trunks)
That concerns me. If the PBX cannot identify the call comes from a trunk then it will not work. What is the outbound proxy? Did you include the transport=tcp parameter in the outbound proxy.
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I am having a lot of problems getting Strato as an ISP for VOIP telephoning to work.
[1] 2008/10/18 23:35:32: Last message repeated 4 times
[7] 2008/10/18 23:35:32: Set packet length to 20
[6] 2008/10/18 23:35:32: Sending RTP for 3c284de6d6f9-ocqyjai495w8#bf058bb268 to 192.168.1.106:60634
[5] 2008/10/18 23:35:32: Dialplan Standard Dialplan: Match 08213240@192.168.1.2 to <sip:08213240@strato-iphone.de;user=phone> on trunk Strato 3
[7] 2008/10/18 23:35:32: Set packet length to 20
[7] 2008/10/18 23:35:32: Call c3db4152@pbx#31941: Clear last INVITE
[5] 2008/10/18 23:35:32: INVITE Response: Terminate c3db4152@pbx
[7] 2008/10/18 23:35:32: Other Ports: 1
[7] 2008/10/18 23:35:32: Call Port: 3c284de6d6f9-ocqyjai495w8#bf058bb268
The Registration works but the handshake falls off. Because of the consolidation of the market here, it is just about impossible to get a tech on the ISP provider side who can understand this issue: Hotlines do not cut it. Everything is set up according to the pbxnsip standard.
I am beginning to think that the ISP Strato is having problems with the packet length. I also opened the Ports from the pbxnsip server I have. The carrier has a different set of values for some of their required port openings, but they also sell another Modem that acts as a Pbx (AVM Fritzbox) as standard for hooking up to their service.
I am at a standstill over how to troubleshoot this. Does anyone have suggestions?
Well strato does not use a session border controller. That is making life difficult. For outbound traffic, you can set the registration time to 30 seconds to keep the NAT bilding alive.
The other problem is that for calls to the PBX the SER sends a UDP packet that exceeds the UDP fragmentation size. The below packet is bigger than 1600 bytes, which will be rejected by most cheap routers.
[7] 2008/10/19 15:47:57: SIP Rx udp:194.97.40.217:5060: INVITE sip:TESTACC@172.23.0.118:5060;transport=udp;line=c4ca4238 SIP/2.0 Record-Route: <sip:194.97.40.217;ftag=3srTHM2ah20004jR0B0Pu003qjW0wKjuX;lr=on> Record-Route: <sip:194.97.96.19;ftag=3srTHM2ah20004jR0B0Pu003qjW0wKjuX;lr=on> Record-Route: <sip:194.97.40.217;ftag=3srTHM2ah20004jR0B0Pu003qjW0wKjuX;lr=on> Via: SIP/2.0/UDP 194.97.54.97;branch=z9hG4bKc916.8304ded2.1;recvip=194.97.40.217 Via: SIP/2.0/UDP 194.97.96.19;branch=z9hG4bKc916.8304ded2.1;recvip=194.97.96.19 Via: SIP/2.0/UDP 194.97.54.97;received=194.97.40.217;branch=z9hG4bK000423D4EB2601AB1EB5D8B227AE;r ecvip=194.97.40.217 Via: SIP/2.0/UDP 194.97.45.169:5060;branch=z9hG4bK000423D4EB2601AB1EB5D8B227AE From: <sip:0019787462777@strato-iphone.de;user=phone>;tag=3srTHM2ah20004jR0B0Pu003qjW0wKjuX To: <sip:004921616371234@strato-iphone.de;user=phone> Call-ID: 000423D4EB2601AB1EB58F5060F3@194.97.45.169 CSeq: 17404 INVITE Contact: <sip:0019787462777@194.97.45.169:5060> Allow-Events: refer Max-Forwards: 14 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE Content-Type: application/sdp Supported: 100rel, timer, replaces User-Agent: TELES.MGC Content-Length: 368 P-Trusted: yes v=0 o=- 1721261723 0 IN IP4 194.97.100.170 s=session c=IN IP4 194.97.100.170 t=0 0 m=audio 30200 RTP/AVP 8 0 18 18 2 36 4 80 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:36 G726-24/8000 a=rtpmap:4 G723/8000 a=rtpmap:80 G723/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv
I am afraid if your NAT router does not support UDP fragmentation you will have no chance to get it working. Even other dirty "tricks" like STUN will not solve this problem.
Maybe someone can give ACME packet, Newport Networks or NexTone a tip there is a good customer waiting for them. Just installing a free SIP proxy does not make you a carrier...
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Thank you. Changing from "never" to "always (no security check)" did the trick !
"Security" should be good by default. I know it causes a lot of grief out there, but if we release a product that happily sends out configuration data with passwords we will get bashed from all kinds of people...
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Can Blind Transfer a call using Xfer + Monitor Button
That is a thing that needs to be fixed in the phone.
Can call another extension by pressing the Monitor buttonThat should work, but you might have to use the BLF mode instead of speed dial.
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I am planning to use an ISP for flatrate calling within my home country and for discounted calling to "landed" phones in neighboring countries. This will be the default carrier. I want to redirect calls to the PSTN (ISDN) trunk only for emergencies and for savings on Mobile phone charges. I have set up the redirection to a Call by Call provider over the PSTN trunk for the Mobile calls and that is working very nicely. I simply put in the exact exchange number such as 017* and the replacement value plus the code and it works wonderfully (01017017*).
There is a problem though. For calls in my hometown I would like to not have to enter the city code all the time. The problem is that City codes vary in length in Germany.
Area codes in Germany consist of 2-5 digits. Usually shorter area codes are used for larger cities, and long area codes are used for smaller towns.
On the other hand the telephone numbers are assigned inversely long. The larger cities have seven or eight digits, while those small cities may have three to four digits. To call a number within the area of Germany dial the number only and to make a call outside Germany one has to use the respective country code.
I am not so experienced that I know which variables do what and would appreciate suggestions.
I would use the preferences in the dial plan. First have a rule for the exceptions, then have a rule for the general. If you
For example:
Pref: 150 Trunk: PSTN Pattern: 017* Replacement: 01017017* Pref: 151 Trunk: PSTN Pattern: 1234567 Replacement: 0401234567 Pref: 152 Trunk: PSTN Pattern: 2345678 Replacement: 0402345678 Pref: 160 Trunk: ITSP Pattern: 00* Replacement: 0101700* Pref: 170 Trunk: PSTN Pattern: 0* Replacement: 0* Pref: 180 Trunk: ISTP Pattern: * Replacement:
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Before the upgrade to version 3 I was able to receive calls to my DID telephone numbers which are Internet based numbers and not ISDN numbers. Since the upgrade I am not getting those calls and I have entered them as both aliases or with a trunk ANI. I need to coordinate the DID to the extention.
The ISP has given 2 3 numbers for internet calling of which 2 are inbound-outbound and one is only outbound. Do I need to set up separate trunks to route the calls inbound or are the aliases enough to do the job. As it is I am not getting the inbound calls on my business line. Ugh.
Did you set the coutry code in the domain settings? Also, if you are using tel:-alias, make sure that your trunk has the "global" flag set.
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I am running versions 3.0.1.3023 and 3.0.0.2998 on two different PBX's. Both systems had Polycom 550's registered to the PBX until I upgraded to 3.0. Now neither system has any of the polycom phones registering. I thought it might be the polycom config, so I updated the bootrom and sip software in the tftp folder, and the phones downloaded the changes (buttons look slightly different), but they still do not register to the PBX. the phone reads "Url call is disabled" when attempting to make a call. Any ideas?
Are you using plug and play? Consider changing the TFTP password policy in the System/Settings/Port section. Maybe the phone just does not get a password provisioned. http://wiki.pbxnsip.com/index.php/Polycom was updated recently
Maybe this would be a good topic to discuss in the Polycom section of this forum...
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it should be 'http://nexsip.com/provisioning/spa$MA.cfg' not '/spa$MA.cfg'
Well the tag "{http-url}" is replaced with something like "http://pbx-adr/provisioning". That should be okay...
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I've succesfully installed a test-setup which contains pbxnsip, ocs 2007 and exchange 2007.
The guide (http://wiki.pbxnsip.com/index.php/Office_Communications_Server) was very usefull.
I'm able to call from one pbxnsip extension to another pbxnsip extension. The call will also be forked to the ocs extension.
Now I like to use Exchange UM as the voicemail system. But when I call a extension and let it ring I won't get Exchange 2007 as a voicemail system.
Exchange UM voicemail works when I call from communicator to a ocs extension.
What did you put into the domain setting "External Voicemail System"?
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I am still having problems with version 3.0.1.3023. it does not allow me to park on any orbit except the one that matches my extension number when using the * in the Explicitly specify park orbit preference field. Are there permissions I am missing, or a newer version available?
Scratch scratch... In that field * has a special meaning... Scratch head scratch... If you put a * there it means that you can only retrieve calls. Don't ask me why, but at the moment that is my understanding (maybe because many phones can only send a simple star code and that's the only way to get some kind of BLF working, at least for call retrieve).
Can't you just list the orbits there?
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I auto provisioned a Linksys PAP2T and noticed that the config file changed the profile rule back to "/spa$MA.cfg" this is bad.
also is the RTP Packet Size: 0.030 right?
What is bad about spa$MA?
<flat-profile> <Profile_Rule ua="na"> {http-url}/spa$MA.cfg </Profile_Rule> <Resync_Periodic ua="na"> 5 </Resync_Periodic> </flat-profile>On the packet size I agree, 30 ms is a bad idea. We'll change that to 20 ms.
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Is there a list of major features that are in V3 that aren't in V2? Just trying to put together a mailer to convince our customers to pay us to upgrade them
Well generally the differences are described in http://wiki.pbxnsip.com/index.php/Release_Notes. That's not a very shiny "markting" document, just facts.
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Works great. Now how can I make the inverse?
Have calls that come in Line 1, ring Extension 100
Have calls that come in Line 2, ring Extension 200
etc
For that you need to set the DID numbers on the PSTN gateway and assign those numbers as alias to extension 100, 200. This will work if you leave the "Send call to extension" field is empty in the trunk. You should then assign all DID numbers to extensions or other accounts, so that all calls get routed somewhere.
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What should I put instead in Replacement?
In that case "sip:2\1@\r;line=2" would be a good replacement (if you want to keep the "2" in the beginning).
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and is it as cool as it looks? I now see a CMC listed as a function on their datasheet.. hmmm
Datasheet looks like it has some extreme integration with pbxnsip.
does RTP multicast paging mean phones now page/intercom from phone to phone and offload the PBX?
Well design is always a question of taste... But the metal foot stand gives a pretty solid impression. Not bad for a device that stands in a high angle. And the display is "crystal" clear. No doubt a great phone.
The integration with pbxnsip is just like the other phones in the 3xx series. Paging and intercom still runs through the PBX. How could intercom be bypassing the PBX?
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is there any reason to nice the pbxnsip to a high priority?
internal to pbxnsip RTP threads take priority i assume?
so if i have 1000 people listening to their voicemail on the web interface, my RTP will not have issues..?
Nice is not enough. The RTP thread needs to run in an different scheduler class. The PBX does that by setting the RTP thread to SCHED_RR. Threads in this scheduler list are processed before any other thread in the "nice" list is being processed.
Yes, SCHED_RR is brutal. Not even the mouse moves when the PBX wants to process RTP.
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How can I configure the CS410 to treat CO lines individually. I mean, for example: Extension 1, only dials out using Line #1. Extension 2 only dials out using Line #2.
You can tell the PSTN gateway what line to use for an outbound call by using the line parameter. For example, when dialling sip:9787462777@1.1.1.1:5060;line=2 the PSTN gateway will use line 2.
Now, in order to use this feature, you need to put that into the replacement part of the dial plan. For example, replacement sip:\1@\r;line=2.
And in order to assign that to a specific user, you need to have a dial plan for every "exception" of the standard dial plan. If you have four users and they should have their own exclusive line, then you need to have four dial plans.
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has pbxnsip been able to pre-test these new snom 820 phones?
yes!
Checking for FAX
in IVR Node Setup
Posted
Yes, got it. We run this through the detector and it does detect the tone. Gain setting was set to 512 ("dtmf_gain"), but I guess that is the same for you (check the pbx.xml file to verify). So the tone detection logic is not the problem here. Did you try this in the auto attendant? Does the auto attendand send the call to the FAX extension? Maybe there is something in the IVR node that is the problem.