My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. I have added "hidecallerid=yes" to zapata.conf and no longer have the problem. But, now I have Legacy PBX users complaining about having no caller ID. I tried this, but it still would not complete the call. hidecallerid=no ;fix for no answer restrictcid=yes usecallingpres=no I have also tried to make the callerid name null, but asterisk still tries to send the data. I have also tried to dumb it down from NI2 to NI1, but asterisk still tries to send the callerID name in the PRI debug. Is there a way to send callerid number and not the name? ref. zapata.conf: context=panasonic swichtype=national pridialplan=unknown prilocaldialplan=unknown signalling=pri_net usecallerid=yes facilityenable=no hidecallerid=yes ;fix for no answer restrictcid=yes usecallingpres=no echocancel=no echocancelwhenbridged=no group=2 channel => 25-47 -- -- Steven http://www.glimasoutheast.org
My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. I have added "hidecallerid=yes" to zapata.conf and no longer have the problem. But, now I have Legacy PBX users complaining about having no caller ID. I tried this, but it still would not complete the call. hidecallerid=no ;fix for no answer restrictcid=yes usecallingpres=no I have also tried to make the callerid name null, but asterisk still tries to send the data. I have also tried to dumb it down from NI2 to NI1, but asterisk still tries to send the callerID name in the PRI debug. Is there a way to send callerid number and not the name? ref. zapata.conf: context=panasonic swichtype=national pridialplan=unknown prilocaldialplan=unknown signalling=pri_net usecallerid=yes facilityenable=no hidecallerid=yes ;fix for no answer restrictcid=yes usecallingpres=no echocancel=no echocancelwhenbridged=no group=2 channel => 25-47 Steven
BerkHolz, Steven
2006-Jun-08 13:58 UTC
[Asterisk-Users] revisit to legacy PBX and CID over PRI
My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. I have added "hidecallerid=yes" to zapata.conf and no longer have the problem. But, now I have Legacy PBX users complaining about having no caller ID. I tried this, but it still would not complete the call. hidecallerid=no ;fix for no answer restrictcid=yes usecallingpres=no I have also tried to make the callerid name null, but asterisk still tries to send the data. I have also tried to dumb it down from NI2 to NI1, but asterisk still tries to send the callerID name in the PRI debug. Is there a way to send callerid number and not the name? ref. zapata.conf: context=panasonic swichtype=national pridialplan=unknown prilocaldialplan=unknown signalling=pri_net usecallerid=yes facilityenable=no hidecallerid=yes ;fix for no answer restrictcid=yes usecallingpres=no echocancel=no echocancelwhenbridged=no group=2 channel => 25-47 Steven
trixter aka Bret McDanel
2006-Jun-08 14:37 UTC
[Asterisk-Users] revisit to legacy PBX and CID over PRI
On Thu, 2006-06-08 at 16:49 -0400, Steven wrote:> My legacy PBX accepts CID number, but not name. > My old PRI vendor never sent the name, so there was never an issue. > > I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. > Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. > The legacy PBX hangs up, but asterisk thinks that it is still ringing. >how long is the caller id string? I believe the spec used on the pstn is 15 characters, while asterisk supports much longer callerid strings that may be a problem. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461 DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060608/d6f4f4ae/attachment.pgp
Kevin P. Fleming
2006-Jun-08 15:11 UTC
[Asterisk-Users] revisit to legacy PBX and CID over PRI
----- Steven <asterisk@tescogroup.com> wrote:> I have also tried to make the callerid name null, but asterisk still > tries to send the data. > I have also tried to dumb it down from NI2 to NI1, but asterisk still > tries to send the callerID name in the PRI debug.There is no option to suppress the calling name being sent... however, libpri currently sends a zero-length DISPLAY information element when you set the calling name to an empty string in Asterisk, which it should not do. Matt Frederickson is fixing that in libpri right now, so you should be able to update in a day or two to the newest libpri code from Subversion (either trunk or branch-1.2, whichever you are running) and solve your problem. If this still does not solve your problem, open a bug at bugs.digium.com with a 'pri debug' trace of the outgoing call to the PBX so we can figure out what is wrong. -- Kevin P. Fleming Senior Software Engineer Digium, Inc.
I have seen this problem with Avaya megix/merlin legend. The workaround is that for every call the goes to the legacy system do something like this: exten => _X.,1,Set(CALLERID(name)"") exten => _X.,2,Dial(whateverconnectstoyourlegacysystem) On 6/8/06, BerkHolz, Steven <StevenBerkHolz@tescogroup.com> wrote:> > My legacy PBX accepts CID number, but not name. > My old PRI vendor never sent the name, so there was never an issue. > > I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI > - asterisk - PRI - Legacy. > Any calls from asterisk (sip and iax extensions) which have callerID > set, will not connect. > The legacy PBX hangs up, but asterisk thinks that it is still ringing. > > I have added "hidecallerid=yes" to zapata.conf and no longer have the > problem. > > But, now I have Legacy PBX users complaining about having no caller ID. > > I tried this, but it still would not complete the call. > hidecallerid=no ;fix for no answer > restrictcid=yes > usecallingpres=no > > I have also tried to make the callerid name null, but asterisk still > tries to send the data. > I have also tried to dumb it down from NI2 to NI1, but asterisk still > tries to send the callerID name in the PRI debug. > > Is there a way to send callerid number and not the name? > > ref. zapata.conf: > > context=panasonic > swichtype=national > pridialplan=unknown > prilocaldialplan=unknown > signalling=pri_net > usecallerid=yes > facilityenable=no > hidecallerid=yes ;fix for no answer > restrictcid=yes > usecallingpres=no > echocancel=no > echocancelwhenbridged=no > group=2 > channel => 25-47 > > Steven > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
mavince@optonline.net
2006-Jun-08 20:17 UTC
[Asterisk-Users] revisit to legacy PBX and CID over PRI
Looking at the bug below, I see an NSF of SDDN> [20 02 00 e6] > Network-Specific Facilities (len= 2) [ ACCUNET Switched Digital Service ]The SDDN NSF is used for digital data, not voice. Try not sending any NSF because an NSF can imply a Call-By-Call trunk group and I have seen that mismatch cause wacky D-channel behavior in the PSTN. If you really need to use an NSF, try SDN. Generally when I see the D-channel restarting, I also check to make sure that there isn't another type of mismatch: Facility Associated Signaling (FAS) on one end and Non-Facility Associated Signaling (NFAS) on the other. Mark Vince Date: Thu, 8 Jun 2006 19:26:14 -0400 From: "C F" <shmaltz@gmail.com> Subject: Re: [Asterisk-Users] revisit to legacy PBX and CID over PRI To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <81000b5a0606081626x287d8e4of8e79650b8fdb35b@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed In paticular with Avaya systems I have seen this problem, it's on the bug tracker here: http://bugs.digium.com/view.php?id=4013