Re: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323 translator
this patch however worked for me, all calls through the patched
chan_h323 are ok, hold, transfer, etc works perfectly. except that there
is no music on hold, while in fact asterisk shows that it is playing,
yet there is no audio heard on the callmanager side.
so i had test on both oh323 0.5.10 and h323(patched) cvs march 20 the
problem with oh323 is that when a call is placed on hold by a
callmanager phone, after resuming, the audio from * to ccm is lagged by
3-4 seconds. while the audio from ccm to * is ok. i already posted this
problem in version 0.5.5. has anybody found a workaround for this?
On Fri, 2004-03-19 at 18:25, Paul Cheng wrote:
Hi,
The patches also did not help us and in fact created some new problems.
The old chan_h323 could pass on early audio and provider messages, but
after the patch, this capability is gone and the channel only rings and
rings while the provider is sending the message.
We've had no problems with the existing chan_h323 other than that it
doesn't return the right indication state to Asterisk, so Asterisk
can't branch for busy versus congestion.
But this is obviously only for our setup.
On Mar 19, 2004, at 9:12 AM, Marian Durkovic wrote:
On Thu, Mar 18, 2004 at 12:22:57PM -0500, Billy Huddleston wrote:
I just tried this, and it's not working for me.. I can't call a 2600
or a
CCM... What version of OpenH323 and PWLIB did you all use?
Are you able to call those without the patches? If not, the patches
won't
help you, since you probably have some other problem..
M.
- Original Message -
From: Marian Durkovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 10:35 AM
Subject: [Asterisk-Users] Several H323 bugfixes - working SIP -
H.323
translator
Hi all,
in an effort to create a SIP - H.323 translator we've found and
fixed
several problems in H.323 channel. These inlcude:
for SIP-H.323 calls
- no ringback tone
- ringback not related to H.323 events
- one-way audio with Cisco CallManager
- incorrect Caller ID
for H.323-SIP calls
- not able to establish call with Cisco IOS 12.3(4)T
- ringback not related to SIP events
- no support for 183 Call Progress
- incorrect Caller ID
Please find the patches against aterisk 0.7.2 release below.
M.
-
-
Marian Durkovic network manager
Slovak Technical University Tel: +421 2 524 51 301
Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351
812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED]
-
-
---
---
Marian Durkovic network manager
Slovak Technical University Tel: +421 2 524 51 301
Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351
812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED]
---
---
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Re: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323 translator
On Thu, Mar 18, 2004 at 12:22:57PM -0500, Billy Huddleston wrote:
I just tried this, and it's not working for me.. I can't call a 2600 or a
CCM... What version of OpenH323 and PWLIB did you all use?
Are you able to call those without the patches? If not, the patches won't
help you, since you probably have some other problem..
M.
- Original Message -
From: Marian Durkovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 10:35 AM
Subject: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323
translator
Hi all,
in an effort to create a SIP - H.323 translator we've found and fixed
several problems in H.323 channel. These inlcude:
for SIP-H.323 calls
- no ringback tone
- ringback not related to H.323 events
- one-way audio with Cisco CallManager
- incorrect Caller ID
for H.323-SIP calls
- not able to establish call with Cisco IOS 12.3(4)T
- ringback not related to SIP events
- no support for 183 Call Progress
- incorrect Caller ID
Please find the patches against aterisk 0.7.2 release below.
M.
--
Marian Durkovic network manager
Slovak Technical University Tel: +421 2 524 51 301
Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351
812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED]
--
--
Marian Durkovic network manager
Slovak Technical University Tel: +421 2 524 51 301
Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351
812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED]
--
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Re: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323 translator
Hi, The patches also did not help us and in fact created some new problems. The old chan_h323 could pass on early audio and provider messages, but after the patch, this capability is gone and the channel only rings and rings while the provider is sending the message. We've had no problems with the existing chan_h323 other than that it doesn't return the right indication state to Asterisk, so Asterisk can't branch for busy versus congestion. But this is obviously only for our setup. On Mar 19, 2004, at 9:12 AM, Marian Durkovic wrote: On Thu, Mar 18, 2004 at 12:22:57PM -0500, Billy Huddleston wrote: I just tried this, and it's not working for me.. I can't call a 2600 or a CCM... What version of OpenH323 and PWLIB did you all use? Are you able to call those without the patches? If not, the patches won't help you, since you probably have some other problem.. M. - Original Message - From: Marian Durkovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 18, 2004 10:35 AM Subject: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323 translator Hi all, in an effort to create a SIP - H.323 translator we've found and fixed several problems in H.323 channel. These inlcude: for SIP-H.323 calls - no ringback tone - ringback not related to H.323 events - one-way audio with Cisco CallManager - incorrect Caller ID for H.323-SIP calls - not able to establish call with Cisco IOS 12.3(4)T - ringback not related to SIP events - no support for 183 Call Progress - incorrect Caller ID Please find the patches against aterisk 0.7.2 release below. M. - - Marian Durkovic network manager Slovak Technical University Tel: +421 2 524 51 301 Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351 812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED] - - --- --- Marian Durkovic network manager Slovak Technical University Tel: +421 2 524 51 301 Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351 812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED] --- --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323 translator
On Fri, Mar 19, 2004 at 11:25:59AM +0100, Paul Cheng wrote: Hi, The patches also did not help us and in fact created some new problems. The old chan_h323 could pass on early audio and provider messages, but after the patch, this capability is gone and the channel only rings and rings while the provider is sending the message. I've not removed the early audio cut-through. For SIP-H.323 direction, the patched version sends: 180 Ringing when Alerting PDU is received from H.323 side 183 Session Progress when asterisk starts getting RTP packets (this is handled in chan_sip.c regardless of H.323 state and I left it untouched) Calls with inband info get both of them i.e. 180 first and 183 afterwards. This might perhaps confuse some SIP clients, but is legal according to RFC3261 and works fine e.g. with Cisco 7940s or Xlite. I'll appreciate any info if this is the problem. The original version never sends 180 Ringing due to various bugs. Thus the SIP caller gets no ringback tone for H.323 calls without inband info. For H.323-SIP direction, the patched version sends: Alerting PDUwhen 180 Ringing received Progress PDU with PI=8 when 183 Session Progress received The original version doesn't detect SIP states and it sends Alerting PDU immediately (even if the user does not exist or is busy). With kind regards, M. -- Marian Durkovic network manager Slovak Technical University Tel: +421 2 524 51 301 Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351 812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323 translator
I just tried this, and it's not working for me.. I can't call a 2600 or a
CCM... What version of OpenH323 and PWLIB did you all use?
- Original Message -
From: Marian Durkovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 10:35 AM
Subject: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323
translator
Hi all,
in an effort to create a SIP - H.323 translator we've found and fixed
several problems in H.323 channel. These inlcude:
for SIP-H.323 calls
- no ringback tone
- ringback not related to H.323 events
- one-way audio with Cisco CallManager
- incorrect Caller ID
for H.323-SIP calls
- not able to establish call with Cisco IOS 12.3(4)T
- ringback not related to SIP events
- no support for 183 Call Progress
- incorrect Caller ID
Please find the patches against aterisk 0.7.2 release below.
M.
--
Marian Durkovic network manager
Slovak Technical University Tel: +421 2 524 51 301
Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351
812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED]
--
___
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Re: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323 translator
Hi all, in an effort to create a SIP - H.323 translator we've found and fixed several problems in H.323 channel. These inlcude: for SIP-H.323 calls - no ringback tone - ringback not related to H.323 events - one-way audio with Cisco CallManager - incorrect Caller ID for H.323-SIP calls - not able to establish call with Cisco IOS 12.3(4)T - ringback not related to SIP events - no support for 183 Call Progress - incorrect Caller ID Please find the patches against aterisk 0.7.2 release below. M. Did you put these files to bugs.digium.com ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323 translator
Will these be available on the CVS? Devel or Stable? Hi all, in an effort to create a SIP - H.323 translator we've found and fixed several problems in H.323 channel. These inlcude: for SIP-H.323 calls - no ringback tone - ringback not related to H.323 events - one-way audio with Cisco CallManager - incorrect Caller ID for H.323-SIP calls - not able to establish call with Cisco IOS 12.3(4)T - ringback not related to SIP events - no support for 183 Call Progress - incorrect Caller ID Please find the patches against aterisk 0.7.2 release below. M. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
