CME CallFwdAll Problem

Answered Question
Mar 31st, 2009
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Hello,


I have a Cisco CME 7.1 system and following problem:


Our calls go out through a G.SHDSL line to a SIP trunk, no other lines (e.g. ISDN).


It is possible to call with an IP Phone (7965G, 8.4.2) an external number (like 06761234567), but if the CallFwdAll is activated, and someone from outside (person A) calls my extension (e.g. 100), then person A gets a busy tone.


The same with the call-forward noan command, extension 100 is reachable, and after the timeout suddenly person A gets a busy tone (number in call-forward noan is configured with 906761234567).


It is a similar problem like http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_tech_note09186a0080816cac.shtml, with the difference that the user is not getting a busy tone in this description.


Call Forwarding is enabled, I configured it like it is explained in the CME Admin Guide.


Does anyone have an idea how this CallFwdAll problem can be solved and with which debug commands I can analyze it? If you need more infos please tell me which you need.


Thanks in advance

Daniel

Correct Answer by exonetinf1nity about 8 years 9 hours ago

Could you try the following please:


Call Forward


When a call comes in on an SIP trunk and gets forwarded (CFNA / CFB / CFA), then the default behavior is for the CME to send the 302 "Moved Temporarily" SIP message to the Service Provider (SP) proxy. The user portion of the Contact Header in the 302 message might need to be translated to reflect a DID that the SP proxy can route to. The host portion of the Contact Header in the 302 message should be modified to reflect the Address of Record (AOR) using the host-registrar CLI under sip-ua and the b2bua CLI under the VoIP dial peer going to the CUE.


Some SIP proxies might not support this. If so, then you need to add this:


Router(config)#voice service voip

Router(conf-voi-serv)#no supplementary-service sip moved-temporarily


Call Transfer


When a call comes in on an SIP trunk to an SCCP Phone or CUE AutoAttendant (AA) and is transferred, the CME by default will send a SIP REFER message to the SP proxy. Most SP Proxy Servers do not support the REFER method. This needs to be configured in order to force the CME to hairpin the call:


Router(config)#voice service voip

Router(conf-voi-serv)#no supplementary-service sip refer



Config Example:


voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer


http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml


HTH


Regards


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Brandon Buffin Tue, 03/31/2009 - 07:41
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Daniel,

Are the incoming calls coming in over the SIP trunk? Do you have the following commands?


voice service voip

allow-connections sip to sip



Hope this helps.


Brandon

shadowhawk3 Tue, 03/31/2009 - 07:46
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Hello Brandon,


yes, incoming calls are coming over the SIP trunk too.


The allow-connections sip to sip command was already configured, this isn't the problem.


Daniel

Brandon Buffin Tue, 03/31/2009 - 07:49
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Does the call forward work if you call from another internal phone?


Brandon

shadowhawk3 Wed, 04/01/2009 - 01:02
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Call forward from inside phone to inside phone is working.

Call forward from inside phone to external number is working.

Call forward from external number to inside phone is working.

Call forward vom external phone to external number is NOT working.


Daniel

Brandon Buffin Wed, 04/01/2009 - 04:52
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Are incoming and outgoing calls on the SIP trunk using the same codec? If not, you'll need a transcoder.


Brandon

shadowhawk3 Wed, 04/01/2009 - 04:56
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Yes, both incoming and outgoing calls use the g711 codec, no transcoding necessary.


Daniel

Brandon Buffin Wed, 04/01/2009 - 05:08
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Are you allowed more than one simultaneous call across the SIP trunk? The scenario that does not work is the only one that has two concurrent calls across the trunk. You might check with your provider to see if they block this functionality as well.


Brandon

Correct Answer
exonetinf1nity Wed, 04/01/2009 - 05:11
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Could you try the following please:


Call Forward


When a call comes in on an SIP trunk and gets forwarded (CFNA / CFB / CFA), then the default behavior is for the CME to send the 302 "Moved Temporarily" SIP message to the Service Provider (SP) proxy. The user portion of the Contact Header in the 302 message might need to be translated to reflect a DID that the SP proxy can route to. The host portion of the Contact Header in the 302 message should be modified to reflect the Address of Record (AOR) using the host-registrar CLI under sip-ua and the b2bua CLI under the VoIP dial peer going to the CUE.


Some SIP proxies might not support this. If so, then you need to add this:


Router(config)#voice service voip

Router(conf-voi-serv)#no supplementary-service sip moved-temporarily


Call Transfer


When a call comes in on an SIP trunk to an SCCP Phone or CUE AutoAttendant (AA) and is transferred, the CME by default will send a SIP REFER message to the SP proxy. Most SP Proxy Servers do not support the REFER method. This needs to be configured in order to force the CME to hairpin the call:


Router(config)#voice service voip

Router(conf-voi-serv)#no supplementary-service sip refer



Config Example:


voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer


http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml


HTH


Regards


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