SIP Line configuration

[brown456]

I also posted this on Stack Overflow but am hoping there are some more Avaya people here:

Hi if anyone else ever needs help connecting an Avaya IP Office phone system with Twilio Elastic SIP trunking, feel free to reach out to me, we got it to work after some trial and error and the connection seems very good. Will save us some good money.

The final piece to the puzzle for us had to do with the IP address that Twilio sends SIP invites from for incoming calls to the Avaya system. To get this to work in the Avaya we had to set up 4 SIP "Lines" pointing to 4 different "ITSP IP Address" values and 4 "Incoming Call Routes" per DID because Twilio sends from 4 different domestic IP addresses. I did read some documentation about being able to leave the ITSP IP Address blank (0.0.0.0) since we do have the "ITSP Domain Name" populated but that does not seem to work in our situation.

In researching this on the Internet I surprisingly did not find much evidence of Avaya customers ever having to deal with this issue (with or without Twilio), however I did find some examples of some other PBX systems having this issue with connecting to Twilio but those systems seemed equipped to configure multiple IP addresses for a single line.


Thus my question - do any Avaya customers have a suggestion for an alternate configuration to make this work? Anything we could do on our firewall? Twilio does not support sending SIP Invites from a single IP address or via a proxy server address. Fortunately we only have about 10 DID's so this won't be that hard to setup and maintain.

[oiduran]

You can configurate IP ITSP or/and Domain. Colud you post the configuration file .cfg of Avaya and the configuration script del twilio?

[zakabog]

Setup the 4 different SIP lines but give the URIs the same incoming group ID, that'll let you use one incoming call route for each DID.

[havel3]

Ever read a doc about SIP trunks on Avaya?
You can have upto 4 IP Adresses for a single provider

ITSP Proxy Address

Default = Blank This is the SIP Proxy address used for outgoing SIP calls. The address can be specified in the following ways:

• If left blank, the ITSP Domain Name is used and is resolved by DNS resolution in the same way as if a DNS address had been specified as below.
• An IP address.
• A list of up to 4 IP addresses, with each address separated by a comma or space.
  
  • The addresses can include an indication of the relative call weighting of each address compared to the others. This is done by adding a w N suffix to the address where N is the weighting value. For example, in the list 213.74.81.102w3 213.74.81.100w2, the weighting values assigns 1.5 times the weight of calls to the first address. The default weight if not specified is 1. A weight of 0 can be used to disable an address. Weight is only applied to outgoing calls.
    If there is more than one proxy defined, and no weight indication, then calls are only sent to the first in the list until there is a failure at which point the next proxy is used.
  • If the Calls Route via Registrar setting below is enabled, the weighting is applied to registrations rather than calls.
• A DNS address, for example sbc.example.com.
  
  • The DNS response may return multiple proxy addresses (RFC 3263). If that is the case, the system will resolve the address to use based on priority, TTL and weighting information included with each address.
  • A load balancing suffix can be added to specify that multiple proxy results should be returned if possible, for example sbc.example.com(N). where N is the required number of addresses from 1 to 4.

This field is mergeable. However, no more than 4 IP Addresses should be in use at any time. So, if the combined new and old address settings exceed 4, the new addresses are only phased into use as transactions in progress on the previous addresses are completed.

[brown456]

Thanks for the reply. I'll track down a file for you. We tried doing 0.0.0.0 for ITSP IP Address with our unique Twilio ITSP Domain in the SIP line config and then none of the calls went through. The INVITES came through but no SIP message was sent back by our server.

You can configurate IP ITSP or/and Domain. Colud you post the configuration file .cfg of Avaya and the configuration script del twilio?

[brown456]

That sounds like a very good idea. I will try that and report back. You are referencing the "SIP URI" tab right that we'd point all 4 of them to line 19 even though we have setup a line 19, 21, 23 and 25, right?

Setup the 4 different SIP lines but give the URIs the same incoming group ID, that'll let you use one incoming call route for each DID.

[brown456]

Thank you for your reply. Is this applicable to IP Office Manager Version 7.0(22)? I only see one block to put in an ITSP IP address for a SIP line and one block for the ITSP Domain Name. Using 0.0.0.0 for the IP address in conjunction w/ the Twilio domain name I had hoped would work but it did not like it.

Ever read a doc about SIP trunks on Avaya?You can have upto 4 IP Adresses for a single provider

ITSP Proxy Address

Default = Blank This is the SIP Proxy address used for outgoing SIP calls. The address can be specified in the following ways:

• If left blank, the ITSP Domain Name is used and is resolved by DNS resolution in the same way as if a DNS address had been specified as below.
• An IP address.
• A list of up to 4 IP addresses, with each address separated by a comma or space.
  • The addresses can include an indication of the relative call weighting of each address compared to the others. This is done by adding a w N suffix to the address where N is the weighting value. For example, in the list 213.74.81.102w3 213.74.81.100w2, the weighting values assigns 1.5 times the weight of calls to the first address. The default weight if not specified is 1. A weight of 0 can be used to disable an address. Weight is only applied to outgoing calls.
    If there is more than one proxy defined, and no weight indication, then calls are only sent to the first in the list until there is a failure at which point the next proxy is used.
  • If the Calls Route via Registrar setting below is enabled, the weighting is applied to registrations rather than calls.
  
  
• A DNS address, for example sbc.example.com.
  • The DNS response may return multiple proxy addresses (RFC 3263). If that is the case, the system will resolve the address to use based on priority, TTL and weighting information included with each address.
  • A load balancing suffix can be added to specify that multiple proxy results should be returned if possible, for example sbc.example.com(N). where N is the required number of addresses from 1 to 4.
  
  

This field is mergeable. However, no more than 4 IP Addresses should be in use at any time. So, if the combined new and old address settings exceed 4, the new addresses are only phased into use as transactions in progress on the previous addresses are completed.

[zakabog]

That sounds like a very good idea. I will try that and report back. You are referencing the "SIP URI" tab right that we'd point all 4 of them to line 19 even though we have setup a line 19, 21, 23 and 25, right?

Exactly, there is the line number and then there's the incoming and outgoing ID, and those numbers don't need to match. So you can have everything share the same incoming or outgoing group ID, it helps keep the ARS and incoming call routes less cluttered.

[brown456]

Exactly, there is the line number and then there's the incoming and outgoing ID, and those numbers don't need to match. So you can have everything share the same incoming or outgoing group ID, it helps keep the ARS and incoming call routes less cluttered.

Yes that works like a charm. Much better, I don't mind setting up 4 SIP lines but wasn't thrilled about dup'ing the other stuff. Thank you thank you.

[pcarlt]

Hi if anyone else ever needs help connecting an Avaya IP Office phone system with Twilio Elastic SIP trunking, feel free to reach out to me

I do need help trying to get this up and running. The main issue is that we can not resolve the ITSP Domain Address field in the Line -> SIP Line -> "ITSP Domain Name" with {our-domain}.pstn.twilio.com

It would be REALLY interesting, step by step, how you would go about getting Twilio to work with 1 SIP trunk on Avaya Office.

Thanks!

[brown456]

I do need help trying to get this up and running. The main issue is that we can not resolve the ITSP Domain Address field in the Line -> SIP Line -> "ITSP Domain Name" with {our-domain}.pstn.twilio.com

It would be REALLY interesting, step by step, how you would go about getting Twilio to work with 1 SIP trunk on Avaya Office.

Thanks!

Yes, this was my issue too. The way we have it working is by creating 4 SIP lines for each of the 4 Twilio domestic IP addresses (SIP Line tab), 54.172.60.0 through 54.172.60.3. And then using zakabog's suggestion we use only one of these line numbers on the "SIP URI" tab. This allows to only have to create 1 set of "Incoming Call Routes" pointing to that same "Line Group ID" as is shown for the "Incoming Group" and "Outgoing Group" on the "SIP URI" tab. Please note that these call routes will require the "+1" for the "Incoming Number", assuming these are US DID's. Feel free to send me a phone number or email address and I'd be glad to arrange a discussion with you or show you my screen.

[hodge46]

I would absolutely love to hear how you've accomplished this! I've been wanting this before twilio even had sip trunking, but unfortunately in my environment I do not get a lot of wiggle room for testing as we only have the one production system. A few screenshots of a sip line would be very much appreciated. Thank you for trying to share your findings.

PS have you tried anything with plivo? I've used both twilio and plivo extensively with freeswitch/asterisk and although I love twilio, I prefer plivo for some things.

[brown456]

I will work on getting you some screenshots together and if you have any specific questions in the meantime let me know. We basically set up four SIP lines for the four different domestic IP addresses that they send calls over. I was not familiar with Plivo. What do you prefer them for?

I would absolutely love to hear how you've accomplished this! I've been wanting this before twilio even had sip trunking, but unfortunately in my environment I do not get a lot of wiggle room for testing as we only have the one production system. A few screenshots of a sip line would be very much appreciated. Thank you for trying to share your findings.

PS have you tried anything with plivo? I've used both twilio and plivo extensively with freeswitch/asterisk and although I love twilio, I prefer plivo for some things.