Voice over IP Cisco - Antw: Analog phones cant transfer calls on 2811

This is Interesting: Free IT Magazines  
Home > Archive > Voice over IP Cisco > September 2007 > Antw: Analog phones cant transfer calls on 2811





You are viewing an archived Text-only version of the thread. To view this thread in it's original format and/or if you want to reply to this thread please [click here]

Author Antw: Analog phones cant transfer calls on 2811
Robert Schuknecht

2007-09-20, 7:11 pm

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

September 2007 um 17:53 in Nachricht 3ab0adee131ef58c7afb0ea50017a410:[vbcol=
seagreen]
> 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
> !
>
>
>
> ---------------------------------
> Be a better Globetrotter. Get better travel answers from someone who knows.
> Yahoo! Answers - Check it out.

Cucumber Green

2007-09-21, 1:11 am

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@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

September 2007 um 17:53 in Nachricht 3ab0adee131ef58c7afb0ea50017a410:[vbcol=
seagreen]
> 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
> !
>
>
>
> ---------------------------------
> Be a better Globetrotter. Get better travel answers from someone who knows.
> Yahoo! Answers - Check it out.




---------------------------------
Don't let your dream ride pass you by. Make it a reality with Yahoo! Autos.
Wes Sisk

2007-09-21, 1:11 am

________________________________________
_______
cisco-voip mailing list
cisco-voip@puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
Ted Nugent

2007-09-21, 1:11 am

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/....html#wp1304455

<http://www.cisco.com/en/US/products....html#wp1049404>





On 9/20/07, Cucumber Green <greencucumber2007@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@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
>
> September 2007 um 17:53 in Nachricht 3ab0adee131ef58c7afb0ea50017a410:
> does
> in
> 4.2(3)
> knows.
>
>
> ------------------------------
> Don't let your dream ride pass you by. Make it a reality<http://us.rd.yahoo.com/evt=51200/*h...YW
1jYXI-
>with Yahoo! Autos.
>
>
> ________________________________________
_______
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


Sponsored Links






Free braindumps | Software forum | Database administration forum

Copyright 2003 - 2008 webservertalk.com