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Tuesday, February 18, 2014

How to Configure and Troubleshoot Call Forward to the PSTN using SIP Trunks ?


All outbound PSTN SIP calls are validated by the SIP Trunk provider to ensure the calls are valid and not toll fraud attempts.  The methods of call validation can vary from provider to provider, and many involve multiple methods for the same call.  This becomes a concern when using the default settings in a CUCM w/CUBE topology and attempting to call forward an inbound PSTN call back out to the PSTN.  One of the most common validation methods is for the SIP provider to examine the "From" field in the incoming INVITE of a call and make sure it matches to a known DID number for that customer.  The default setting in CUCM for forwarding calls is to maintain the CLID of the calling originator.  This causes the "From" field of an outgoing INVITE to contain the CLID number of the original PSTN caller, and not a valid DID belonging to the customer.  When the provider sees this with no additional information, it typically will reject the call setup attempt or ignore it completely.

There are three common solutions to this issue. 

The first is to alter the "From" field so that the CUCM will send the information of the redirecting number instead of the call originator.  In this scenario the "From" field will contain a valid customer DID number, and the call will validate.  The advantage of this technique is that it works with virtually all SIP providers.  The disadvantage is that the receiving party of the forwarded call will only see the caller-ID of the number that was set to forward (typically their own office DID number).  They will not see the caller-ID of the original caller, and therefore will not know who is calling prior to answering the call. 

Here are the steps to use the first method:
-Navigate to the SIP Trunk page in CUCM
-Navigate to "Outbound Calls" section of the page
-Change the value in the drop down entitled "Calling Party Selection" to "Last Redirect Number (External)"
-Save the change, and apply the change (may drop calls why you press apply, so be careful with this change during business hours)

A second method is also often used, though it requires the SIP provider to validate calls through a different method.  This technique involves adding a p-asserted identity line in the outbound INVITE containing a valid DID.  The advantage of this technique is that the "From" field retains the calling originator's information so that the recipient of the call can see who is calling on their caller-ID.  The disadvantage of this technique is that it is not universally used, and customers need to check with their SIP providers to make sure they will validate calls based on the p-asserted identity value instead of the "from" field value. 

Here are the steps to use the second technique when using a CUBE gateway:
-Log into the CUBE router
-Add the following SIP profile (make sure to use a unique profile number if you have existing SIP profiles, or add it to your outbound SIP profile in use.  Also make sure to use a valid DID number and SBC URL information.)

voice class sip-profiles 1
request INVITE sip-header P-Asserted-Identity add "P-Asserted-Identity:1111111111@sbc.sipprovider.com
>"

-Add the following configuration line to the dial-peer being matched to place the outbound call to the SIP provider:

voice-class sip profiles 1

Make sure that the SIP Profile contains a valid DID with your SIP provider.  Also, most providers will look for a fully qualified domain name in the PAI, as opposed to an IP address.  Be use to use the FQDN in the profile, even if you are routing the calls directly to the provider's IP address.

The third method is similar to the second but instead of a PAI line, this uses a Diversion Header.  The Diversion Header is added to the outbound INVITE and contains a DID number that the provider can validate the call with.  Like with the second option, the advantage of this technique is that the "From" field retains the calling originator's information so that the recipient of the call can see who is calling on their caller-ID.  The disadvantage of this technique is that it is not universally used, and customers need to check with their SIP providers to make sure they will validate calls based on the diversion header value instead of the "from" field value. 

Here are the steps to use the third technique when using a CUBE gateway:
-Navigate to SIP Trunk page in CUCM
-Navigate to "Outbound Calls" section of the page
-Check the "Redirecting Diversion Header Delivery - Outbound" Check box
-Save the change, and apply the change (may drop calls why you press apply, so be careful with this change during business hours)
-Log into the CUBE router
-Add the following SIP profile (make sure to use a unique profile number if you have existing SIP profiles, or add it to your outbound SIP profile in use.  Also make sure to use a valid DID number and SBC URL information.)

voice class sip-profiles 1
request INVITE sip-header Diversion modify "" "1111111111@sbc.sipprovider.com
>"

-Add the following configuration line to the dial-peer being matched to place the outbound call to the SIP provider:

voice-class sip profiles 1

Make sure that the SIP Profile contains a valid DID with your SIP provider.  Also, most providers will look for a fully qualified domain name in the diversion header, as opposed to an IP address.  Be use to use the FQDN in the profile, even if you are routing the calls directly to the provider's IP address.

Troubleshooting
The following debugs are useful in this call scenario:
debug ccsip messages
debug voip ccapi inout
When reading these debugs keeping the different call legs straight can get confusing.  It is recommended to use an application such as Notepad ++, and to mark the Call ID of each call leg traversing the CUBE.  The goal of these debugs is to check that the outbound INVITE for the forwarded call contains either the correct "from" field, or the correct "p-asserted identity" or "diversion header" field.  If using the "from" field method, check that the SIP INVITE coming from the CUCM contains the full 10 digit DID number for the last redirecting party, and that this value is the same in the INVITE going to the provider.  If using either the "p-asserted identity" method or the "diversion header" method, make sure that the outgoing call is matching the correct dial-peer that has the voice-class sip profile set on it.  If it is matching a different dial-peer, add the sip profile to that dial-peer.

Citation - This blog post does not reflect original content from the author. Rather it summarizes content that are relevant to the topic from different sources in the web. The sources might include any online discussion boards, forums, websites and others.

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