Please help check my ARS OUT Pattern

star8888 · Post by **star8888** » Wed Aug 06, 2008 2:50 am

1. Brekeke Product Name and version: Brekeke PBX

2. Java version:

3. OS type and the version:

4. UA (phone), gateway or other hardware/software involved:

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :

6. Your problem: Hi All, I want to have One Stage Dialing, below is my Patterns - OUT in PBX please help to check is it correct? I still not able to do a one stage dialing....

Patterns - OUT
Matching To sip:600(.*)@

Note: 600 is Prefix

Deploy patterns sip:$1@192.168.1.3 -> my FXO IP address

I want make a call to PSTN just dial 600 then PSTN no#(eg. 107) instead of dial 600 -> hear tone -> dial PSTN no#

Please advise the Pattern OUT

Thanks.

hope · Post by **hope** » Wed Aug 06, 2008 9:40 am

you need to setup one stage dialing at FXO side

hope · Post by **hope** » Wed Aug 06, 2008 11:17 am

or try using ARS, here 600 is FXO number registered in SIP server

Patterns - OUT
Matching patterns:
To: sip:600(.+)@

Deploy patterns:
To: sip:600
DTMF: $1#

star8888 · Post by **star8888** » Wed Aug 06, 2008 6:54 pm

Hi Hope, THANKS a lot. It's work. I am able to call from IP Phone to PBX extension without hear the dial tone!

I have one more question here, is possible for me no need dial the 600? Mean I just need to call the phone no# without dial the fxo prefix (600)?

Now 600(prefix) 123456(phone no#)

Possible ignore the 600(prefix) using the ARS in Brekeke PBX?

star8888 · Post by **star8888** » Thu Aug 07, 2008 1:07 am

Hi Hope, Just now I have posted another new issue. Sorry to check with you again. I have problem call from PSTN to my IP phone.

I am using one of my office PSTN desk phone call to my IP phone, dial 108 (GW FXS port) hear tone then dial 2000 (IP Phone no#), but can not, IP phone no ring at all.

Do I need to do any setting in ARS? Please advise.

hope · Post by **hope** » Thu Aug 07, 2008 1:39 pm

I have one more question here, is possible for me no need dial the 600? Mean I just need to call the phone no# without dial the fxo prefix (600)?

Now 600(prefix) 123456(phone no#)

if all pstn phone numbers have 6 digits, such as 123456
change the ARS Matching patterns as
To: sip:(.{6})@

now any dialing number with exact 6 digits will be sent to FXO

star8888 · Post by **star8888** » Mon Aug 11, 2008 7:34 pm

Hi Hope, need your help again.

I have below for One-Stage-Dialing-Patterns - OUT

Matching partterns:
To sip:9(.+)@

Deploy partterns:
To sip:9
DTMF $1#

* SIP call to PSTN Extension
===> dial 9(fxo prefix) 107(extension no#)

* SIP call to external (Mobile phone)
====> dial 9(fxo prefix) 9(pstn prefix) 123456(mobile no#)

To remove the 9(fxo prefix), are you asking me change the above Matching patterns to ==> To: sip:(.{6})@ ?

How about the Deploy patterns? Do I need to change?

Please advise. Thanks.

hope · Post by **hope** » Tue Aug 12, 2008 11:08 am

* SIP call to PSTN Extension
===> dial 9(fxo prefix) 107(extension no#)

* SIP call to external (Mobile phone)
====> dial 9(fxo prefix) 9(pstn prefix) 123456(mobile no#)

in this case, try
Patterns - OUT
Matching patterns:
To: sip:(1|9)(.+)@
//here (1|9) is used to match all pstn extensions start with 1,
//and use prefix 9 to dial external
//it is better the pstn extensions prefix is different from pbx users prefix

Deploy patterns:
To: sip:9
DTMF: $1$2#

star8888 · Post by **star8888** » Wed Aug 13, 2008 7:41 pm

Hi Hope, Thanks. It's work!!! We have facing another issue this morning.

sip call to sip (our sip phone using wireless), during the call we notice that there are some occasional jittery in the voice quality when we speak to each other.

I did try to change the Audio Codec in sip phone to G.729a,
G.711-U and G.711-A, result still the same.

Is there a setting in SIP server for me to adjust? Any idea?

star8888 · Post by **star8888** » Thu Aug 14, 2008 2:09 am

Hi Hope, I would like to provide more information here,

Note: we are using wifi phone (connected to AP)

PSTN Ext call to wifi phone voice quality is ok, quite good.

BUT

When wifi phone to wifi phone, voice quality is different:-
wifi phone (#206) call to wifi phone (#207) ----> jitter in the voice quality.

I have tired call to the IP Address, eg. wifi phone (#206) call to another wifi phone dial the IP address 207@10.1.1.46, voice quality ok also. No problem.

Hope above information is enough for youo to understand our situation.

Please advise is there a setting in Brekeke?

hope · Post by **hope** » Thu Aug 14, 2008 5:04 pm

set pbx/options/setting/PBX system settings/rtp relay: off
or
set pbx/options/setting/PBX system settings/rtp relay:on and also from each phones user edit page in pbx, set "rtp relay": on

to see which way the sounds get better
set codec G.711-U to both wireless phones

star8888 · Post by **star8888** » Fri Aug 15, 2008 1:49 am

Hi Hope, ok I will test.

Between, How about the echo? Can adjust in Brekeke?

Please advise.

star8888 · Post by **star8888** » Fri Aug 15, 2008 2:59 am

Hi Hope,

We did below:-

The better result is set the rtp replay:off in PBX and G.711-U in both wireless phone. But voice quality still bad, voice like "Robot" talking - h...e....l....l...o.....

Our wireless phone do not have rtp setting so I am not able to set it to rtp ON.

Any idea what can we do?

How about other PBX system settings? (eg. Codec Priority) can help?

Anyway LAN extension to wireless phone or wireless phone to LAN extension voice is very good and clear. Just have an echo....

lakeview · Post by **lakeview** » Mon Aug 18, 2008 10:36 am

Can you capture RTP packets?
Let you that check RTP packets are exchanged between wireless phones directly without the PBX.
If you can see RTP packets on the PBX machine, it means RTP packets are still relayed.

star8888 · Post by **star8888** » Mon Aug 18, 2008 10:18 pm

Hi Lakeview, I am asking our engineer to capture the RTP packets.

We use the Brekeke SIP server do not have this voice issue. Is there a different between Brekeke SIP Server and Brekeke PBX?

lakeview · Post by **lakeview** » Tue Aug 19, 2008 7:39 pm

>> the Brekeke SIP server do not have this voice issue.

It seems that RTP packets are still relayed via the PBX's machine.

Please make sure "RTP-Relay = off" at both Brekeke PBX and Brekeke SIP Server.

>> Is there a different between Brekeke SIP Server and Brekeke PBX?

It can be..
Because the PBX converts codecs if both clients' codecs are different.

star8888 · Post by **star8888** » Tue Aug 19, 2008 9:24 pm

Hi Lakeview,

I did set the Brekeke SIP Server Configuration > RTP -> RTP exchanger > RTP reply: auto (tried On also) and RTP replay (UA on this machine): OFF.

Brekeke PBX -> Options -> Settings -> PBX system settings -> RTP relay : OFF

Codec in both wifi phone set to G.711-U and also tried G.729a and G.711-A (both using same codec).

Still same result, the voice quality not improved at all.

Any suggestion what can we resolve this issue? Thanks.

lakeview · Post by **lakeview** » Tue Aug 19, 2008 10:34 pm

As I posted..
Try capturing of RTP packets.
And check packets are exchanged between wireless phones directly without the PBX.

star8888 · Post by **star8888** » Wed Aug 20, 2008 12:19 am

Hi lakeview, yes our engineer will do it. Thanks for your help.