SIP phone without using local dial plan

All-, Does anyone know any parameter available on Cisco 7940/7960 which start using the switch dial plan instead of phone specific dial plan ? The protocol used here is SIP. Feby Francis

Normally, the SIP endpoint that device that collects digits is responsible for knowing when a correct and complete number pattern has been dialed. The same is true for MGCP devices when configured to match a digitmap; they only send off the NTFY when a pattern of digits has been dialed, or else a timeout has occurred. I'm almost certain that there is no way in the Cisco 7940/7960 SIP firmware to override this norm. You probably need to configure the digit map in the phone to match the digit map from your switch. However, some platforms will let you do something similar: when you pick up the phone, you can have the CPE automatically INVITE to a specific SIP uri, like sip:switch at telco.com Then the switch could accept that media stream, play back the local dialtone, and collect digits itself. A similar method is used on platforms like BroadWorks and Metaswitch to provide the "second dial tone" some users expect to hear after dialing 9 "for an outside line". I can't recommend for this approach, under normal circumstances. There are a lot of advantages in having the SIP endpoint construct a proper SIP INVITE using the digits dialed, with a sip...;user=phone URI. mark at ecg.co | +1-229-316-0013 | http://ecg.co/lindsey On Sep 23, 2011, at 09:21 , Feby Francis wrote:
Does anyone know any parameter available on Cisco 7940/7960 which start using the switch dial plan instead of phone specific dial plan ? The protocol used here is SIP.

On 23-Sep-11 09:08, Mark R Lindsey wrote:
Normally, the SIP endpoint that device that collects digits is responsible for knowing when a correct and complete number pattern has been dialed. The same is true for MGCP devices when configured to match a digitmap; they only send off the NTFY when a pattern of digits has been dialed, or else a timeout has occurred.
I'm almost certain that there is no way in the Cisco 7940/7960 SIP firmware to override this norm. You probably need to configure the digit map in the phone to match the digit map from your /switch/.
That's the normal expectation. However, if the client and server correctly implement the 484 (Address Incomplete) error code, that is not required. Here's how it /should/ work: 1. User dials "9" 2. Client sends INVITE for sip:9 at server (or tel:9) 3. Server responds with 484 4. Client waits for more digits 5. User dials "1" 6. Client sends INVITE for sip:91 at server (or tel:91) 7. Server responds with 484 8. Client waits for more digits 9. User dials "1" 10. Client sends INVITE for sip:911 at server (or tel:911) 11. Server completes call normally I have no idea whether the Cisco 79xx code does this, but it should be relatively easy to test if your server is known to support it: just give the phone a digit map that accepts any single digit and see what happens. S -- Stephen Sprunk "God does not play dice." --Albert Einstein CCIE #3723 "God is an inveterate gambler, and He throws the K5SSS dice at every possible opportunity." --Stephen Hawking

-----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Stephen Sprunk Sent: Friday, September 23, 2011 4:41 PM To: voiceops at voiceops.org Subject: Re: [VoiceOps] SIP phone without using local dial plan On 23-Sep-11 09:08, Mark R Lindsey wrote: Normally, the SIP endpoint that device that collects digits is responsible for knowing when a correct and complete number pattern has been dialed. The same is true for MGCP devices when configured to match a digitmap; they only send off the NTFY when a pattern of digits has been dialed, or else a timeout has occurred. I'm almost certain that there is no way in the Cisco 7940/7960 SIP firmware to override this norm. You probably need to configure the digit map in the phone to match the digit map from your switch. That's the normal expectation. However, if the client and server correctly implement the 484 (Address Incomplete) error code, that is not required. Here's how it should work: 1. User dials "9" 2. Client sends INVITE for sip:9 at server (or tel:9) 3. Server responds with 484 4. Client waits for more digits 5. User dials "1" 6. Client sends INVITE for sip:91 at server (or tel:91) 7. Server responds with 484 8. Client waits for more digits 9. User dials "1" 10. Client sends INVITE for sip:911 at server (or tel:911) 11. Server completes call normally I have no idea whether the Cisco 79xx code does this, but it should be relatively easy to test if your server is known to support it: just give the phone a digit map that accepts any single digit and see what happens. ---------------------------------------- REPLY -------------------- It has been a while, but I think Ciscos (SIP) phones support some kind of "hotline" (aka batphone) feature. If your PBX has a DISA feature you can hotline to DISA to get a dialtone and have the PBX collect digits. This is an ugly hack. The method above is the "right way", but if your phone does not support 484, this method might work. Be VERY careful to make sure DISA cannot be accessed from outside your PBX.
participants (4)
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EWieling@nyigc.com
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feby.francis@crosstel.com
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lindsey@e-c-group.com
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stephen@sprunk.org