Greetings, I have a problem making a call from Asterisk to Cisco H323 PSTN gateway using H323 channel. I can call but there are no sound in both way. If I call H323 gateway directly from SJPhone I have no problem with sound. Any advice are welcome. Thanks in advance.
Cisco and Asterisk are not behind firewall.
Where can I check for settings noH245Tuneling and noFastStart in Asterisk
H323?
-----
-- Executing Dial("SIP/msn-069a",
"H323/73952389506@peer:1720") in new stack
-- Called 73952389506@peer:1720
-- H323/peer:1720 is making progress passing it to SIP/msn-069a
-- H323/peer:1720 is ringing
-- H323/peer:1720 answered SIP/msn-069a
== Spawn extension (messanger, 73952389506, 1) exited non-zero on
'SIP/msn-069a'
--------------
-----Original Message-----
From: ht@phonitel.com [mailto:ht@phonitel.com]
Sent: Monday, February 14, 2005 11:29 AM
To: Vitalie.Apostu@Compuflex.Net
Subject: Re: [Asterisk-Users] Asterisk-H323
Make sure settings for:
noH245Tuneling and noFastStart parameters are correctly tuned both sides.
Is Cisco or Asterisk behind NAT?
Send more info
>
> Greetings,
>
> I have a problem making a call from Asterisk to Cisco H323 PSTN
> gateway using H323 channel. I can call but there are no sound in both
> way. If I call
> H323 gateway directly from SJPhone I have no problem with sound.
>
> Any advice are welcome.
>
> Thanks in advance.
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
noH245Tunneling instead of noH245Tuneling
typedef struct call_options {
char cid_num[80];
char cid_name[80];
int noFastStart;
int noH245Tunneling;
int noSilenceSuppression;
unsigned int port;
int progress_setup;
int progress_alert;
int progress_audio;
int dtmfcodec;
} call_options_t;
-----Original Message-----
From: ht@phonitel.com [mailto:ht@phonitel.com]
Sent: Monday, February 14, 2005 11:29 AM
To: Vitalie.Apostu@Compuflex.Net
Subject: Re: [Asterisk-Users] Asterisk-H323
Make sure settings for:
noH245Tuneling and noFastStart parameters are correctly tuned both sides.
Is Cisco or Asterisk behind NAT?
Send more info
>
> Greetings,
>
> I have a problem making a call from Asterisk to Cisco H323 PSTN
> gateway using H323 channel. I can call but there are no sound in both
> way. If I call
> H323 gateway directly from SJPhone I have no problem with sound.
>
> Any advice are welcome.
>
> Thanks in advance.
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
hi, im trying to add H323 capability of asterisk to communicate with my VOIP gateway which is H323 compliant. but it seems no luck for me to have error free binary build with my machine. I downloaded the latest aserisk-oh323-0.7.1 and the required libraries (Openh323 1.13.5, pwlib 1.6.6). is there any version mismatch or path needed to have a succesful build? i got an error when i done MAKE to the asterisk-oh323. anybody have tried to a succesfull build...i badly need your hand on this... k0l0s0s Philippines __________________________________ Do you Yahoo!? All your favorites on one personal page – Try My Yahoo! http://my.yahoo.com
Hi
I've this plattaform
sip sjphone - - - asterisk- - -gatekeeper- - -ohphone- - -phonejack card-
- analog phone
Asterisk is registered with Gatekeeper
Ohphone is registered with Gatekeeper
Phonejack is installed successfully and get a dial tone
When i try to call phonejack with sip phone i have this message :
call with mysjphone ('alias/IP' of asterisk) completed duration 0:00
and analog phone don't ring
when i try to call mysjphone with analog phone i've this message:
Speed Dial 3231 not defined ,trying gatekeeper..
phonejack is calling host 3231
'alias' 'ip' of asterisk is busy duration 0:01
Have you suggestions?
Thanks Ale