[cisco-voip] Antw: Analog phones cant transfer calls on 2811acting asH323 gw to UCM 4.2 (3)
Lelio Fulgenzi
lelio at uoguelph.ca
Fri Sep 21 00:01:29 EDT 2007
perhaps...
----- Original Message -----
From: Ted Nugent
To: Lelio Fulgenzi
Cc: Cucumber Green ; cisco-voip at puck.nether.net
Sent: Thursday, September 20, 2007 11:44 PM
Subject: Re: [cisco-voip] Antw: Analog phones cant transfer calls on 2811acting asH323 gw to UCM 4.2 (3)
Yep same here and I wondered about that also until i tried to add an ISR FXS SCCP port to CM 4.1... its not an option like in 5.1 for example, I'm guessing thats what they're referring to...?
On 9/20/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
I think the documentation is misleading...I'm using supplementary services w/ SCCP on the VG224 with CCM 4.1(3) with no problems.
----- Original Message -----
From: Ted Nugent
To: Cucumber Green
Cc: cisco-voip at puck.nether.net
Sent: Thursday, September 20, 2007 10:21 PM
Subject: Re: [cisco-voip] Antw: Analog phones cant transfer calls on 2811acting asH323 gw to UCM 4.2 (3)
If you need features with fxs you might want to look into SCCP for FXS controller as you would with a VG224 and h323 for and PSTN/PRI access etc. (see the link you need CM4.2+ and a certain rev of IOS depending on your hardware).
http://www.cisco.com/en/US/partner/products/ps6441/products_feature_guide09186a0080483a76.html#wp1304455
On 9/20/07, Cucumber Green <greencucumber2007 at yahoo.com > wrote:
Hello!
Yeah I have tried this without luck, I guess it only works if the H323 gateway has a FXO
connection to a PBX
Robert Schuknecht <rschuknecht at gmx.de> wrote:
Hi Cucumber,
did you try to adjust the the hook-flash timer under the voice-port?
This timer has to equal on the voice-port and the connected telephone.
/Robert
>>> Cucumber Green schrieb am Donnerstag, 20.
September 2007 um 17:53 in Nachricht 3ab0adee131ef58c7afb0ea50017a410:
> Analog phones can send and receive calls from IP phones, but
> when pressing the hookflash key at the analog phone, the analog phone does
> not get a dial tone in order to transfer the call :(
>
> Phones are hooked into FXS ports on a 2811 and will use G729 codec.
>
> I have already checked the "Media Termination Point required" check box in
> the gateway configuration at call manager (using unified call manager 4.2(3)
> ). I thought that would do the trick, but it didnt, any ideas?
>
> This is the gateway config (IOS version 12.4(3g) spservicesK9)
>
> !
> version 12.4
> service timestamps debug datetime msec
> service timestamps log datetime msec
> service password-encryption
> !
> ip subnet-zero
> !
> !
> ip cef
> !
> !
> no ip domain lookup
> !
> voice-card 0
> no dspfarm
> !
> !
> !
> !
> interface Loopback0
> ip address X.X.X.X 255.255.255.255
> h323-gateway voip bind srcaddr X.X.X.X
> !
> !
> ip classless
> ip route 0.0.0.0 0.0.0.0 X.X.X.X 254
> !
> !
> control-plane
> !
> !
> !
> voice-port 0/1/0
> signal loopStart
> !
> voice-port 0/1/1
> signal loopStart
> !
> dial-peer voice 10 pots
> destination-pattern 50001
> port 0/1/0
> !
> dial-peer voice 20 pots
> destination-pattern 50002
> port 0/1/1
> !
> dial-peer voice 1 voip
> preference 1
> destination-pattern 1...
> session target ipv4:X.X.X.X
> dtmf-relay h245-alphanumeric
> codec g729r8 bytes 40
> ip qos dscp cs3 media
> ip qos dscp cs3 signaling
> !
> dial-peer voice 4 voip
> preference 2
> destination-pattern 1...
> session target ipv4:X.X.X.X
> dtmf-relay h245-alphanumeric
> codec g729r8 bytes 40
> ip qos dscp cs3 media
> ip qos dscp cs3 signaling
> !
>
>
>
> ---------------------------------
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