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Sip Trunks & Co-ordinated Dialing Plan
- Thread starter canflyguy
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- #2
I'm just playing around with the SIP Trunking on the BCM50 and I see I can register multiple SIP accounts, but I'm trying to figure out settings for how to link two or more BCMs together? I see I can put up a Private account, but then it refuses the entry as the outbound SIP trunks have 0 1 2 3 4 5 6 7 8 9 listed in the routing table for destination digits and it gives me the entry "Dialed Digits string overlaps with an existing entry"?
I'm trying to figure out if the same SIP trunks since they can be used with multiple SIP providers, can also route between BCMs private network?
I'm trying to figure out if the same SIP trunks since they can be used with multiple SIP providers, can also route between BCMs private network?
curlycord
Programmer
0 1 2 3 4 5 6 7 8 9 listed"
Poor programming!
Remove all that.
If you set up a Dest code of 91 then put in 91 in the SIP Routing table too.
In my system I have Dest codes 91 to 99 for SIP lines, some 8X's and 7XX's for other things.
So what ever is in your Dest Code list that uses a Route/BlockA then make sure the SIP Routing table has the same entries.
Also Absorb should = 0
Instead let the SIP Trunking/Accounts do the Absorbing, 2 in my case.
I did a detailed FAQ on this, have a look.
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Toronto, Canada
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Poor programming!
Remove all that.
If you set up a Dest code of 91 then put in 91 in the SIP Routing table too.
In my system I have Dest codes 91 to 99 for SIP lines, some 8X's and 7XX's for other things.
So what ever is in your Dest Code list that uses a Route/BlockA then make sure the SIP Routing table has the same entries.
Also Absorb should = 0
Instead let the SIP Trunking/Accounts do the Absorbing, 2 in my case.
I did a detailed FAQ on this, have a look.
________________________________________
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Toronto, Canada
Add me to LinkedIN
- Thread starter
- #4
Thanks Curlycord!
I'll give it a try in a little bit. So if the other routes are private routes to access other BCMs, what do you enter for the destination digits, domain and port? I had a 3 on destination digits as I thought it was the number of digits I was passing to the other BCM?
I'll give it a try in a little bit. So if the other routes are private routes to access other BCMs, what do you enter for the destination digits, domain and port? I had a 3 on destination digits as I thought it was the number of digits I was passing to the other BCM?
- Thread starter
- #5
I tried putting the same 9 number in both routing tables as you suggested, (using only a 9) but found that it put the 9 out in the outgoing number and the route is specified as a national. As an example 001 regular outgoing voip using 9 as the outgoing with no absorb and then putting 9 in the sip route. The BCM monitor shows 9 going out with the phone number so I don't where you'd set it to absorb the 9? Also, it didn't seem to stop at 10 digits. I could put out 15 if I kept dialing additional digits.
I was then going to use 002 for the BCM to BCM route and specified it as Private. Once I got the regular dialing to go out on the Public route 001.
I was then going to use 002 for the BCM to BCM route and specified it as Private. Once I got the regular dialing to go out on the Public route 001.
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- #6
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- #7
I've played around with this a bit and although I've got the calls to go out on the Public SIP account, there doesn't seem to be a limit or cutoff at the 10 digits and it has to do interdigit time-out to process the call. Also on the Private SIP, it doesn't stop the calls at 3 digit passthrough. It seems to allow for as many as you can hammer out before it tries to process the call. I haven't put up the other BCM I'm connecting to yet.
curlycord
Programmer
Telephony/Dialing Plan/Public Network/ Public Network DN Lenghts
Try adding your dest code there too, set the amount of digits and if needed the timeout in Telephony/Dialing Plan/General
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Toronto, Canada
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Try adding your dest code there too, set the amount of digits and if needed the timeout in Telephony/Dialing Plan/General
________________________________________
=----(((((((((()----=
Toronto, Canada
Add me to LinkedIN
- Thread starter
- #9
Hi Curlycord!
I've messed around with the settings for a while now and although I can at times get the calls to go out on the public network, it seems somehow inconsistent. I've set up DID's and PRI's and SIP trunks for a while now with no problem. This time is a bit different as I'm trying to use the same line pool of SIP trunks to both go out to the SIP provider and as well link them (by routing (if possible)), to go out to another BCM or 2 or 3. I've got the public and private DN lengths set for 10 and 3 and when either is set up in the routing tables individually, they work fine. When I fight to enter both by way of either my initial attempts or when placing 9 or 91 and 92 in the routing tables and the SIP trunks with the absorb set there to get rid of the leading digit(s), the call does not go out correctly or consistently as the digit lengths don't seem to take effect. It also reflects the leading digit (9 at the moment) in the BCM Monitor when testing calls. Unless (and I'm guessing) that I have to also edit the public Network DN Lengths table to reflect the leading digit(s) that we're using for an access code? Am I right to think that you can actually combine the SIP trunk pool to service both Public and Private routes on the same line pool? I know I can easily oombine analogue or PRI with SIP?
Thanks in advance!
I've messed around with the settings for a while now and although I can at times get the calls to go out on the public network, it seems somehow inconsistent. I've set up DID's and PRI's and SIP trunks for a while now with no problem. This time is a bit different as I'm trying to use the same line pool of SIP trunks to both go out to the SIP provider and as well link them (by routing (if possible)), to go out to another BCM or 2 or 3. I've got the public and private DN lengths set for 10 and 3 and when either is set up in the routing tables individually, they work fine. When I fight to enter both by way of either my initial attempts or when placing 9 or 91 and 92 in the routing tables and the SIP trunks with the absorb set there to get rid of the leading digit(s), the call does not go out correctly or consistently as the digit lengths don't seem to take effect. It also reflects the leading digit (9 at the moment) in the BCM Monitor when testing calls. Unless (and I'm guessing) that I have to also edit the public Network DN Lengths table to reflect the leading digit(s) that we're using for an access code? Am I right to think that you can actually combine the SIP trunk pool to service both Public and Private routes on the same line pool? I know I can easily oombine analogue or PRI with SIP?
Thanks in advance!
curlycord
Programmer
Unless (and I'm guessing) that I have to also edit the public Network DN Lengths table to reflect the leading digit(s) that we're using for an access code?"
That is exactly what I said in my last answer.
I have only Networked sites twice.
These are my hand made notes from my last one if some of it helps, I had lost my original notes from the first one.
These two sites already had the same DN range 2XX when they were Norstars
They dial 7X2XX (Destcode&DN)to call each other...not that often.
Notes:
This guide is to network BCM's via SIP VoIP Trunks
SIP or Gateway Trunks key codes are required.
-The following is based on programming site A using Dest code 71 and Route 071 to call site B (or Dest 72/Route 072 to call site C and so on)
-Others sites that want to call site A would use Dest code 70 and Route 070
To make a call you dial the dest code followed by the DN (7X2XX)
-All sites have the same DN ranges (221- 2XX) in this setup
SIP Trunking/IP Trunk Settings
-Last Redirect
-Remote MWI - yes
-Send name Display - yes
SIP Trunking/Private/Routing Table
Name of site- Site2
Destination Digits - 7X
Domain - localdomain or theirdomain.xxx (DNS domain name
IP address - external public ip address (or internal if in same network)
Port - 0
GW Type - BCM
MCDN - CSE
---You will need to add a table for each site except itself
SIP Trunking/Media Parameters
Make all 3 codecs available
Telephony/Global/FeatureSettings/Business Names
-Program a Name, make sure to put Space after the name
Telephony/Sets
-Program OLI as 70221
-Allow BlocA in Line Access
Telephony/Lines/Target Lines
-Private Received - 2 digit Dest plus 3 digit DN (ie 70221)
-Prime Set - None
-Assign a set (I use Ring Only and min two intercom keys to free up buttons)
-Check Caller ID
Telephony/Lines/Active Voip Lines
-Remove Prime set from Line 001, it will then remove from rest of voip lines too.
This will prevent all 7XXXX entries ringing 221, if you do not do this then Received Digits in Target Lines will appear to be ignored.
Telephony/Dialing Plan/Private Network
-Private Recieved - 3
-Private DN Lenght - 3
CDP
ID 1
Telephony/Dialing Plan/Routes/
-Route 070:
Use Pool - BlocA
DN type - Private
Telephony/Dialing Plan/Dest Codes
-Dest code 70
Normal Route - 070
Absorb Lenght - 0
----To Dial out using POT's lines on remote site A (head office)
Branch Office - Site B
-Private Network type = Recieved and DN Lenght = 3, CDP, ID #1
-Dest code 9, Absorb All, Use Route 009
-Route 009, External #9, BlocA, Public
SipTrunking/Private - Routing Table 9 out 9 localdomain 192.168.1.XXX (siteA) Port 0 - GW Type BCM - MCDN Protocol CSE - Monitor false(0) - Tx Threshold 0.0
Voip Trunks use Remote Package 05
Remote Package 05 has access to BlocA and Pool A
Head Office - Site A
-Private Network type = Recieved and DN Lenght = 3, CDP, ID #1
-Dest code 9A, Absorb All, Use Route 000
-Route 000, use Pool A
-Public Netowork - Add Pub Rec Digits lenght maybe if it was grabbing a Voip Trunk
________________________________________
=----(((((((((()----=
Toronto, Canada
Add me to LinkedIN
That is exactly what I said in my last answer.
I have only Networked sites twice.
These are my hand made notes from my last one if some of it helps, I had lost my original notes from the first one.
These two sites already had the same DN range 2XX when they were Norstars
They dial 7X2XX (Destcode&DN)to call each other...not that often.
Notes:
This guide is to network BCM's via SIP VoIP Trunks
SIP or Gateway Trunks key codes are required.
-The following is based on programming site A using Dest code 71 and Route 071 to call site B (or Dest 72/Route 072 to call site C and so on)
-Others sites that want to call site A would use Dest code 70 and Route 070
To make a call you dial the dest code followed by the DN (7X2XX)
-All sites have the same DN ranges (221- 2XX) in this setup
SIP Trunking/IP Trunk Settings
-Last Redirect
-Remote MWI - yes
-Send name Display - yes
SIP Trunking/Private/Routing Table
Name of site- Site2
Destination Digits - 7X
Domain - localdomain or theirdomain.xxx (DNS domain name
IP address - external public ip address (or internal if in same network)
Port - 0
GW Type - BCM
MCDN - CSE
---You will need to add a table for each site except itself
SIP Trunking/Media Parameters
Make all 3 codecs available
Telephony/Global/FeatureSettings/Business Names
-Program a Name, make sure to put Space after the name
Telephony/Sets
-Program OLI as 70221
-Allow BlocA in Line Access
Telephony/Lines/Target Lines
-Private Received - 2 digit Dest plus 3 digit DN (ie 70221)
-Prime Set - None
-Assign a set (I use Ring Only and min two intercom keys to free up buttons)
-Check Caller ID
Telephony/Lines/Active Voip Lines
-Remove Prime set from Line 001, it will then remove from rest of voip lines too.
This will prevent all 7XXXX entries ringing 221, if you do not do this then Received Digits in Target Lines will appear to be ignored.
Telephony/Dialing Plan/Private Network
-Private Recieved - 3
-Private DN Lenght - 3
CDP
ID 1
Telephony/Dialing Plan/Routes/
-Route 070:
Use Pool - BlocA
DN type - Private
Telephony/Dialing Plan/Dest Codes
-Dest code 70
Normal Route - 070
Absorb Lenght - 0
----To Dial out using POT's lines on remote site A (head office)
Branch Office - Site B
-Private Network type = Recieved and DN Lenght = 3, CDP, ID #1
-Dest code 9, Absorb All, Use Route 009
-Route 009, External #9, BlocA, Public
SipTrunking/Private - Routing Table 9 out 9 localdomain 192.168.1.XXX (siteA) Port 0 - GW Type BCM - MCDN Protocol CSE - Monitor false(0) - Tx Threshold 0.0
Voip Trunks use Remote Package 05
Remote Package 05 has access to BlocA and Pool A
Head Office - Site A
-Private Network type = Recieved and DN Lenght = 3, CDP, ID #1
-Dest code 9A, Absorb All, Use Route 000
-Route 000, use Pool A
-Public Netowork - Add Pub Rec Digits lenght maybe if it was grabbing a Voip Trunk
________________________________________
=----(((((((((()----=
Toronto, Canada
Add me to LinkedIN
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Hi Curlycord!
I'm playing around here again today, but wondered if maybe I'm trying to achieve the impossible? I'm trying to use the same SIP trunk group to process both regular outgoing calls to everywhere as well as using the same SIP trunk line pool to route "private" to other BCMs. I don't know if that's possible or not. I also am not sure what the difference is between putting 9 or 9A as outgoing destination digits? When I play with some of the settings that you're suggesting, I don't get any local default 10 digit cut-off and the same with the 3 digit cutoff for the private routes?
Is what I'm trying possible?
I'm playing around here again today, but wondered if maybe I'm trying to achieve the impossible? I'm trying to use the same SIP trunk group to process both regular outgoing calls to everywhere as well as using the same SIP trunk line pool to route "private" to other BCMs. I don't know if that's possible or not. I also am not sure what the difference is between putting 9 or 9A as outgoing destination digits? When I play with some of the settings that you're suggesting, I don't get any local default 10 digit cut-off and the same with the 3 digit cutoff for the private routes?
Is what I'm trying possible?
curlycord
Programmer
I recommend using 2 digits for Dest codes.
A means Any
What is a SIP Trunk Group?
To call out on SIP you program the Routes, Dest, DN Length etc in Dialing Plan and then in SIP Trunks Public program Routing etc
To call site B you program the Routes, Dest, DN Length etc in Dialing Plan and then in SIP Trunks Private Routing etc
________________________________________
=----(((((((((()----=
Toronto, Canada
Add me to LinkedIN
A means Any
What is a SIP Trunk Group?
To call out on SIP you program the Routes, Dest, DN Length etc in Dialing Plan and then in SIP Trunks Public program Routing etc
To call site B you program the Routes, Dest, DN Length etc in Dialing Plan and then in SIP Trunks Private Routing etc
________________________________________
=----(((((((((()----=
Toronto, Canada
Add me to LinkedIN
- Thread starter
- #14
Hi Curlycord!
After some experimentation, I finally got the two systems which are working here on the same subnet talking to each other. Both can access outside calls. Both can call back and forth to each other. I wanted to test this before trying to implement it outside of here. The only problem and one that is obviously major and significant is that I don't have audio between the two systems. I don't know if that's because it's on the same network? Signalling works and transfers work, except no audio once you've answered the call?
I'm not sure why?
After some experimentation, I finally got the two systems which are working here on the same subnet talking to each other. Both can access outside calls. Both can call back and forth to each other. I wanted to test this before trying to implement it outside of here. The only problem and one that is obviously major and significant is that I don't have audio between the two systems. I don't know if that's because it's on the same network? Signalling works and transfers work, except no audio once you've answered the call?
I'm not sure why?
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- #15
Follow up. I found that because both systems were both registered to the same SIP provider is why the audio was not working. I changed the sources to different IP addresses for the SIP provider and all is good. I've tested audio calls back and forth and everything works fine including transferred calls. The only thing left is that when transferring calls back and forth they were initially tromboning. Then I enabled TAT, but now whenever I try to transfer back it says. Trunk 12 hung up and won't allow a transfer. I'm going to reboot both and see if that clears it. I'll let you know.
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