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Type-B SIP phone delay after pressing "Dial"

Hi *,

I noticed that phone does not make a call instantly when I enter digits and then press "Dial" or pickup handset. It still waits for interdigit timer (T302) to expire. Is it possible to avoid this delay and send digits "en block" in this case?

This shouldn't be mixed with situation when handset is picked up and then digits are entered.

Thanks!

Dragan

Everyone's tags (1)
8 REPLIES 8
Highlighted
Rising star

You can check "urgent

You can check "urgent priority" to route the call immediatley on matching route pattern.

here is a good explanation how it works.

https://supportforums.cisco.com/discussion/12156151/urgent-priority

Highlighted

Hi Deepak,

Hi Deepak,

Unfortunately, this wouldn't be a solution here, because of dial plan with various number of digits. Also, the situation is not present with SCCP phones, only SIP...

Thanks anyway!

Dragan

Highlighted
Rising star

SRND -10.X

SRND -10.X

How urgent patterns can be used to used to avoid intergigit timeout for variable lenght numbers.

It is clearly mentioned it is possible to avoid interdigit timeout by using urgent translation pattern and subsequent route pattern.

Also Last  option will be too decrease the  t302 timer value.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/dialplan.html

Highlighted
Beginner

Hi Dragan,

Hi Dragan,

I am facing the same issue. would you have any solution so far??

Highlighted
Beginner

https://supportforums.cisco

https://supportforums.cisco.com/document/87236/working-concept-sccp-sip-phones-and-dial-rules

Type B SIP Phones: Cisco IP Phone models 79x1, 79x2, 79x5, 7970 and 7960

Type B and new model phones supports KPML and SIP dial rules. KPML is turned on by default on all Type B phone models 79x1, 79x2, 79x5, 7970 and7960

Type B SIP Phones: No Dial Rules

Cisco Type B SIP IP Phones offer functionality based on the KPML standard to report user activities. Each user input event (dialed digit or softkey/button) generates a KPML message to CUCM. This mode of operation emulates a similar end-user experience to that of phones using SCCP.

 

Type B SIP Phones: Dial Rules

Cisco Type B SIP Phones offer functionality based SIP INVITE Message. Every key the end user presses triggers an individual SIP message. The first event is communicated with a SIP INVITE, but subsequent messages use SIP NOTIFY messages. The SIP NOTIFY messages send KPML events corresponding to any buttons or soft keys pressed by the user. Cisco Type B SIP IP Phones with SIP dial rules operate in the same manner as Cisco Type A phones with dial rules

Highlighted
Cisco Employee

TYPE-B SIP Phone behavior

TYPE-B SIP Phone behavior with or w/o Dial-rule on CUCM.


when a users pressing a digit form his/her keypad sip notify msg send every event to the CUCM. then CM perform the digit-digit analysis(DA)
to recognize the patterns as the user dials them. once the route pattern match is found in CUCM then call manager provides a call progress indication either by ringback or re-order after dialing last digit without having to press the dial softkey

where as in Dial-rule scenario , Digit are process against the SIP dial-rules which is configured on CM and decide fi the call can be permitted then the digits are sent enbloc to CUCM using SIP INVITE without the need t0 press the Dial key If a dialed pattern does not match a SIP Dial Rule, the user has to wait for inter-digit timeout before the SIP NOTIFY message is sent to CUCM. Users do not need Dial softkey clicked to indicate the end of dialing But, when on-hook dialing is used, users can click Dial soft key at any time to trigger sending of all dialed digits to CUCM in one SIP INVITE message If a particular pattern is permitted by the phone but blocked by CUCM, the user must dial the entire dial string before receiving an indication that the call is rejected by the system

I am not certain whether you have configured the dial-rules or not but there are some other workaround to fix this issue. like decrease the T302 timer or make specific match of destination number

This has explain very well in this  doc

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/5x/50dialpl.html#wp1090080

Br

Nadeem

Br, Nadeem Please rate all useful post.
Highlighted

Hi Nadeem,

Hi Nadeem,

I tried with configuring SIP dial rules and applying them to some of test phones, but T.302 delay is still there when I enter digits (while on hook) and pressing "Dial". The same happens when hanging up after entering digits. This does not correspond to behavior of SCCP phones...

For example, I tried with SIP dial pattern  "0...." and dialing 01234, but the phone still waits for more digits. Do you have some idea where if I am doing something wrong here?

Thanks!

Dragan

Highlighted
Beginner

Be careful with "!"

Please be aware about the following two bugs: 

- CSCup99586 "Delay during enbloc dialing from SIP endpoints and Jabber client", and 

- CSCuy47914 "enbloc dialing T302 delay BU2BU discussion case".

 

Both the bugs deal with the same problem - SIP phones do not send any indication to CUCM when dialing is completed. Thereby, CUCM is capable to route a call only at the moment when a received number exactly matches to some pattern/DN, or interdigit timeout shall occur. 

 

This behavior is seen in all the cases, neglecting the method used to indicate/detect the end of a dialing process, e.g. we see it when user presses "Call" softkey, goes off-hook, and when the dialed number matches to the SIP dial rule. As result, both the "Call" softkey and SIP Dial Rules have no any impact and are absolutely useless.

 

In my opinion it is a serious defect inside SIP phones, as phone shall inform CUCM immediately when user presses "Call" softkey, or when SIP Dial Rule is matched. Strange that the bug has been rated as severity 6 issue (enhancement request), especially in situation when many SCCP phones are replaced with SIP phones, and users expect to get the same experience as before.

 

Unfortunately, the only workaround is to escape "!" as much as possible, but it is not realistic in multi-country setups, as well all the route patterns has to be revised, and many specific rules added.

 

P.S. In reality, SIP phones indicate end of a dialing process to CUCM by sending a KPML message "Timer Expired", but this happens only at the moment when interdigit timer expires. So, both phone and CUCM just wait for the same, and waste our time...

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