CCME 4 Conference via SIP Trunk, DSPs

Answered Question
Apr 29th, 2008
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Hi everyone,


I've got a couple of questions, so please bare with me.


1) I'm performing a call via a configured SIP trunk, to an overseas number, using my CCME 4.0.3 system that runs on a 2811. At the same time, I'd like to perform a conference by dialing another number (local), but when I hit the 'Conference button', I receive the following message on my 7975G ip phone:

Cannot Complete Conference


I haven't configured any dspfarm on my router, so I'd like to know if you able to use DSP's for transcoding calls that use SIP trunks.


2) On another note, I've got a pvdm2-32 installed (32 channels), but when I give the "show voice dsp" command, I receive only the following output:


----------------------------FLEX VOICE CARD 0 ------------------------------

*DSP VOICE CHANNELS*


CURR STATE : (busy)inuse (b-out)busy out (bpend)busyout pending

LEGEND : (bad)bad (shut)shutdown (dpend)download pending


DSP DSP DSPWARE CURR BOOT PAK TX/RX

TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT

===== === == ========= ======= ===== ======= === == ========= == ==== ============

*DSP SIGNALING CHANNELS*

DSP DSP DSPWARE CURR BOOT PAK TX/RX

TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT

===== === == ========= ======= ===== ======= === == ========= == ==== ============

C5510 001 05 {flex} 9.4.4 alloc idle 0 0 0/1/0 02 0 36/0

C5510 001 06 {flex} 9.4.4 alloc idle 0 0 0/1/1 06 0 36/0

C5510 001 07 {flex} 9.4.4 alloc idle 0 0 0/1/2 10 0 36/0

C5510 001 08 {flex} 9.4.4 alloc idle 0 0 0/1/3 14 0 36/0

------------------------END OF FLEX VOICE CARD 0 ----------------------------


Shouldn't I be seeing more DSPs on this card ? I'm expecting 8 DSP's (8x4=32), but can only see 4. Any ideas ?



3) Lastly, I've configured a dial-peer voip that sends all '00T' numbers via the sip trunk previously mentioned. When I give the "show dialplan number 0061298889990" command, I receive the following :


Macro Exp.: 0061298889990

No match, result=1


But the call is processed via the sip trunk normally. Does the "show dialplan number xxxx" only work for POTS dial peers ?


Many thanks in advanced !

Correct Answer by paolo bevilacqua about 9 years 1 month ago

Hi,


really you don't need to worry about DSPs. You have enough and that's all what really matters. Anyway, the reason why g.729 is shown is because it's the default for a dialpeer until another codec gets selected, possibly you never used ephone-dn 2, 67, etc. And the things numbered 1...40 are channels for ephone-dn, these don't consume any DSP resoruce unless the call goes via a PSTN port.


Here is how you strip 9 when sending a call to a ITSP.


voice translation-rule 99

rule 1 /^9/ //


voice translation-profile strip9

translate called 99


dial-peer voice 99 voip

destination-pattern 9.....T

translation-profile outgoing strip9

...etc...



Note if you choose your extension number wisely, you can do with the '9' and without stripping, eg all extensions start with '2'. All other numbers are called "as-is" via pstn or ITSP.


Hope this helps, please rate post if it does!

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paolo bevilacqua Thu, 05/01/2008 - 07:10
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Hi,


1. for usable hardware conferencing, please upgrade to a minimum of CME 4.1 - IOS 12.4(11)XJ4. Previous versions requires the use of two E1/T1 port in loopback. All the details are in "CME system administrator guide", under "configuring conferencing"


2. the DSP output s correct. What is telling you, is that a single DSP (numbered 001) is handling four ports. The PVDM2-32 has 2 DSPs and each one can process up to 16 calls.


3. Add the keyword "timeout" at the end of "show dialplan" command, to match DPs that have T in destination-pattern.


Hope this helps, please rate post if it does!

cpartsenidis Thu, 05/01/2008 - 14:26
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Bevilacqua,


Many thanks for your reply. Can we further clarify the following? :


Regarding Q No2:

The full output of 'show voice dsp' is the following


2811#show voice dsp

DSP DSP DSPWARE CURR BOOT PAK TX/RX

TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK COUNT

==== === == ======== ========== ===== ======= === == ========= == ===== ============

edsp 001 01 g711ulaw 0.1 IDLE 50/0/1.1

edsp 002 02 g711ulaw 0.1 IDLE 50/0/1.2

edsp 003 01 g711ulaw 0.1 IDLE 50/0/2.1

edsp 004 02 g729r8 p 0.1 IDLE 50/0/2.2

......

edsp 030 02 g729r8 p 0.1 IDLE 50/0/62.2

edsp 031 01 g729r8 p 0.1 IDLE 50/0/63.1

edsp 032 02 g729r8 p 0.1 IDLE 50/0/63.2

edsp 033 01 g729r8 p 0.1 IDLE 50/0/64.1

edsp 034 02 g729r8 p 0.1 IDLE 50/0/64.2

edsp 035 01 g729r8 p 0.1 IDLE 50/0/65.1

edsp 036 02 g729r8 p 0.1 IDLE 50/0/65.2

edsp 037 01 g729r8 p 0.1 IDLE 50/0/66.1

edsp 038 02 g729r8 p 0.1 IDLE 50/0/66.2

edsp 039 01 g729r8 p 0.1 IDLE 50/0/67.1

edsp 040 02 g729r8 p 0.1 IDLE 50/0/67.2


----------------------------FLEX VOICE CARD 0 ------------------------------

*DSP VOICE CHANNELS*


DSP DSP DSPWARE CURR BOOT PAK TX/RX

TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT

===== === == ========= ======= ===== ======= === == ========= == ==== ============

*DSP SIGNALING CHANNELS*

DSP DSP DSPWARE CURR BOOT PAK TX/RX

TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT

===== === == ========= ======= ===== ======= === == ========= == ==== ============

C5510 001 05 {flex} 9.4.4 alloc idle 0 0 0/1/0 02 0 36/0

C5510 001 06 {flex} 9.4.4 alloc idle 0 0 0/1/1 06 0 36/0

C5510 001 07 {flex} 9.4.4 alloc idle 0 0 0/1/2 10 0 36/0

C5510 001 08 {flex} 9.4.4 alloc idle 0 0 0/1/3 14 0 36/0

------------------------END OF FLEX VOICE CARD 0 ----------------------------


Can you tell me why I have a total of 40 edsp's (shouldn't I have 32?) and why the 'codec' column has a mixture of g711ulaw/g729 ?


Also, the output shows 1 DSP allocated to my 4 port FXO card, but I also have an additional VIC2-2BRI-NT/TE card. Shouldn't I be seeing something similar for the VIC2-2BRI-NT/TE card ?


Regarding Q No.3:

When using a dial-peer voip, is there any way we can make use of commands such as 'forward-digits all', 'prefix' e.t.c to help manipulate the final number that will be sent to the SIP provider ?

The problem is that I need to distinguish when the system will place a call using SIP (by using a '9'). I created a dial-peer voip with the destination-pattern number 9T, but I don't want the '9' to be sent to the sip provider!



Many thanks,


Correct Answer
paolo bevilacqua Thu, 05/01/2008 - 15:08
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Hi,


really you don't need to worry about DSPs. You have enough and that's all what really matters. Anyway, the reason why g.729 is shown is because it's the default for a dialpeer until another codec gets selected, possibly you never used ephone-dn 2, 67, etc. And the things numbered 1...40 are channels for ephone-dn, these don't consume any DSP resoruce unless the call goes via a PSTN port.


Here is how you strip 9 when sending a call to a ITSP.


voice translation-rule 99

rule 1 /^9/ //


voice translation-profile strip9

translate called 99


dial-peer voice 99 voip

destination-pattern 9.....T

translation-profile outgoing strip9

...etc...



Note if you choose your extension number wisely, you can do with the '9' and without stripping, eg all extensions start with '2'. All other numbers are called "as-is" via pstn or ITSP.


Hope this helps, please rate post if it does!

cpartsenidis Wed, 05/07/2008 - 00:31
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Dear Paolo,


Thanks again for your informative response!


I've got one more question regarding the translation rules:


Assume I have the following 4 MSN numbers on my phone line connected to the BRI interface of my CCME router:


500500

500501

500502

500503


Because 500503 is my fax number, I want to configure the system so that it does not answer that specific number, therefore the fax will pick it up after 3 or 4 rings.


Could you provide an example of such a voice translation profile and how I can bind it to the voice port?


Many thanks,

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