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] > > > > > > > > --- > > --- > > ___ > > 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 ___ 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
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 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
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
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
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] > > -- > ___ 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] 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] -- --- chan_h323.h.072 Tue Jan 13 09:46:46 2004 +++ chan_h323.h Thu Mar 18 16:03:11 2004 @@ -84,6 +84,7 @@ function*/ typedef struct call_options { char *callerid; + char *callername; int noFastStart; int noH245Tunnelling; int noSilenceSuppression; @@ -101,6 +102,7 @@ const char *call_dest_alias; const char *call_source_e164; const char *call_dest_e164; + const char *call_source_name; const char *sourceIp; } call_details_t; @@ -134,6 +136,11 @@ typedef void (*start_logchan_cb)(unsigned int, const char *, int); start_logchan_cb on_start_logical_channel; +/* This is a callback prototype function, called when openh323 + OnAlerting is invoked */ +typedef void (*chan_ringing_cb)(unsigned); +chan_ringing_cbon_chan_ringing; + /* This is a callback protoype function, called when the openh323 OnConnectionEstablished is inovked */ typedef void (*con_established_cb)(unsigned); @@ -167,6 +174,7 @@ on_connection_cb, start_logchan_cb, clear_con_cb, + chan_ringing_cb, con_established_cb, send_digit_cb); @@ -189,6 +197,8 @@ /* H323 create and destroy sessions */ int h323_make_call(char *host, call_details_t *cd, call_options_t); int h323_clear_call(const char *); + int h323_send_alerting(const char *token); + int h323_send_progress(const char *token); int h323_answering_call(const char *token, int); int h323_soft_hangup(const char *data); --- chan_h323.c.072 Tue Jan 13 10:24:26 2004 +++ chan_h323.c Thu Mar 18 16:09:40 2004 @@ -388,7 +389,7 @@ int res; struct oh323_pvt *p = c->pvt->pvt; char called_addr[256]; - char *tmp; + char *tmp, *cid, *cidname, oldcid[256]; strtok_r(dest, "/", &(tmp)); @@ -419,15 +420,47 @@ /* Copy callerid, if there is any */ if (c->callerid) { - char *tmp = strchr(c->callerid, '"'); - if (!tmp) { - p->calloptions.callerid = malloc(80); // evil - // sprintf(p->calloptions.callerid, "\"%s\"", c->callerid); - sprintf(p->calloptions.callerid, "\"\" <%s>", c->callerid); - } else { - p->calloptions.callerid = strdup(c->callerid); - } -} +memset(oldcid, 0, sizeof(oldcid)); +memcpy(oldcid, c->callerid, strlen(c->callerid)); +oldcid[sizeof(oldcid)-1] = '\0'; +ast_callerid_parse(oldcid, &cidname, &cid); +if (p->calloptions.callerid) { +free(p->calloptions.callerid); +p->calloptions.callerid = NULL; +} +if (p->calloptions.callername) { +free(p->calloptions.callername); +p->calloptions.callername = NULL; +} +p->calloptions.callerid = (char*)malloc(256); +if (p->calloptions.callerid == NULL) { +ast_log(LOG_ERROR, "Not enough memory.\n"); +return(-1); +} +memset(p->calloptions.callerid, 0