I see there's a SetCallerID app I can call; sorry didn't see that until after I sent the mail. If anyone else is interested: exten => 725,1,SetCallerID,725 exten => 725,2,Dial,SIP/1(cellnumber)@iconnect As has been discussed already - IConnect works relatively well and doesn't tie up a line for the outgoing. If that fails I can fall through to a dial by Zap, but that's not a concern as voicemail is always an option. Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers BitShop, Inc. - http://www.bitshop.com - $149/month colo special -----Original Message----- From: Steve Radich [mailto:stever at bitshop.com] Sent: Saturday, February 22, 2003 5:58 PM To: 'asterisk-users at lists.digium.com' Subject: [Asterisk-Users] Override Caller ID? I'm working on a solution for myself to give different people different extensions to reach me; I'm off site quite a bit and want these extensions to fwd to my cell phone when I have call forwarding on at my desk. I want to change the caller id sent to reflect the extension dialed, or a specific caller id - NOT the original callers caller id - i.e. I want to config in my extensions.conf a Dial/.../callerid=123-456 Is there a way to do this? It looks relatively easy to patch Dial to do it, however I'm not sure I follow where all the caller id stuff is being stored/retrieved from and want to make sure I'm not patching something that can already be done. Thanks, Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers BitShop, Inc. - http://www.bitshop.com - $149/month colo special -----Original Message----- From: Steve Radich [mailto:stever at bitshop.com] Sent: Saturday, February 22, 2003 3:13 PM To: 'asterisk-users at lists.digium.com' Subject: [Asterisk-Users] Agressive Echo Cancel Problem.. First let me say the new aggressive echo cancel seems to work wonders. However in testing I tried a transfer and when I pressed flash on the phone the caller experienced a horrible squealing sound they said. I transferred to another phone I could reach, hit flash again to join the calls and heard this noise myself on the new line joined in - The original line I didn't hear it (or I may have just failed that section of the hearing test <grin>). Anyone else experiencing this? Other than this transfer issue the new echo cancel sounds great, Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers BitShop, Inc. - http://www.bitshop.com - $149/month colo special -----Original Message----- From: Klaus-Peter Junghanns [mailto:kpj at junghanns.net] Sent: Saturday, February 22, 2003 2:58 PM To: asterisk-users at lists.digium.com Subject: Re: [Asterisk-Users] inband DTMF in RTP hi mark, what's so ugly about this idea? i have modified chan_sip to support inband dtmf. it's configurable in sip.conf on a per peer basis. regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET Internet-Services & Software-Development GmbH Breite Strasse 13 - 12167 Berlin - Germany fon: +49 30 79705392 fax: +49 30 79705391 mobile: +49 160 7503372 email: kpj at junghanns.net Am Sam, 2003-02-22 um 20.35 schrieb Mark Spencer:> it could be patched to do so but this is an ugly idea in general. > > Mark > > On Thu, 20 Feb 2003, Ben Clark wrote: > > > Is it possible to configure asterisk to understand inband DTMF duringSIP calls?> > > > > > --------------------------------- > > Do you Yahoo!? > > Yahoo! Tax Center - forms, calculators, tips, and more > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ Asterisk-Users mailing list Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
Does this work? I've tried this same config with no luck... Each time I call out from asterisk on iconnect it is a "private number" on the remote caller id. On Saturday, February 22, 2003, at 05:52 PM, Steve Radich wrote:> I see there's a SetCallerID app I can call; sorry didn't see that until > after I sent the mail. > > If anyone else is interested: > > exten => 725,1,SetCallerID,725 > exten => 725,2,Dial,SIP/1(cellnumber)@iconnect > > As has been discussed already - IConnect works relatively well and > doesn't > tie up a line for the outgoing. If that fails I can fall through to a > dial > by Zap, but that's not a concern as voicemail is always an option. > > Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers > BitShop, Inc. - http://www.bitshop.com - $149/month colo special > > > -----Original Message----- > From: Steve Radich [mailto:stever at bitshop.com] > Sent: Saturday, February 22, 2003 5:58 PM > To: 'asterisk-users at lists.digium.com' > Subject: [Asterisk-Users] Override Caller ID? > > I'm working on a solution for myself to give different people different > extensions to reach me; I'm off site quite a bit and want these > extensions > to fwd to my cell phone when I have call forwarding on at my desk. > > I want to change the caller id sent to reflect the extension dialed, > or a > specific caller id - NOT the original callers caller id - i.e. I want > to > config in my extensions.conf a Dial/.../callerid=123-456 > > Is there a way to do this? > > It looks relatively easy to patch Dial to do it, however I'm not sure I > follow where all the caller id stuff is being stored/retrieved from > and want > to make sure I'm not patching something that can already be done. > > Thanks, > > Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers > BitShop, Inc. - http://www.bitshop.com - $149/month colo special > > > -----Original Message----- > From: Steve Radich [mailto:stever at bitshop.com] > Sent: Saturday, February 22, 2003 3:13 PM > To: 'asterisk-users at lists.digium.com' > Subject: [Asterisk-Users] Agressive Echo Cancel Problem.. > > First let me say the new aggressive echo cancel seems to work wonders. > > However in testing I tried a transfer and when I pressed flash on the > phone > the caller experienced a horrible squealing sound they said. I > transferred > to another phone I could reach, hit flash again to join the calls and > heard > this noise myself on the new line joined in - The original line I > didn't > hear it (or I may have just failed that section of the hearing test > <grin>). > > Anyone else experiencing this? > > Other than this transfer issue the new echo cancel sounds great, > > Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers > BitShop, Inc. - http://www.bitshop.com - $149/month colo special > > > -----Original Message----- > From: Klaus-Peter Junghanns [mailto:kpj at junghanns.net] > Sent: Saturday, February 22, 2003 2:58 PM > To: asterisk-users at lists.digium.com > Subject: Re: [Asterisk-Users] inband DTMF in RTP > > hi mark, > > what's so ugly about this idea? i have modified chan_sip > to support inband dtmf. it's configurable in sip.conf on a > per peer basis. > > regards > kapejod > > -- > Klaus-Peter Junghanns > > CEO,CTO > Junghanns.NET Internet-Services & Software-Development GmbH > Breite Strasse 13 - 12167 Berlin - Germany > fon: +49 30 79705392 > fax: +49 30 79705391 > mobile: +49 160 7503372 > email: kpj at junghanns.net > > > Am Sam, 2003-02-22 um 20.35 schrieb Mark Spencer: >> it could be patched to do so but this is an ugly idea in general. >> >> Mark >> >> On Thu, 20 Feb 2003, Ben Clark wrote: >> >>> Is it possible to configure asterisk to understand inband DTMF during > SIP calls? >>> >>> >>> --------------------------------- >>> Do you Yahoo!? >>> Yahoo! Tax Center - forms, calculators, tips, and more >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users at lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
Perhaps your problem is the telco that is completing your call. I have been playing with this myself and have a couple of little observations for calls I placed through Asterisk via iconnect: 1. When I dial my cell phone, I see whatever caller ID I set in Asterisk (with a 1 prepended), regardless of how invalid the number is (e.g., 411). If I don't set it, I see 1 plus my 8-digit iconnect account number. 2. When I dial my landline, I see a specific number (in area code 646) no matter what I set in Asterisk. Perhaps the telco substitutes the ANI number when the caller ID it receives doesn't match? I don't really know. 3. Same as (2) when I dialled my sister's cell, which is on a different network than mine. Cheers, Brad On Mon, 24 Feb 2003 00:43:27 0 (GMT), Ben Clark wrote:> > This is how I have it... I don't understand why it will > not work for > me. Anyone have ideas on what I could try? > > exten => _1NXXNXXXXXX,1,SetCallerID,"Asterisk > <3128847514>" > exten => _1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION at deltathree > > On Sunday, February 23, 2003, at 12:30 AM, Shawn > Djernes wrote: > > > to get that to work you need to write it like. > > > > exten => 725,1,SetCallerID "Name Here <8005551212>" > > > > Note if you are calling to PSTN only the number will > be transfered > > > > > > > > -- > > Shawn L. Djernes > > shawn at djernes.org | sdjernes at telerama.com | > sdjernes at earthlink.net > > <ahref="http://mail.mail1.emailfiltering.co.uk/jump/http://www.djernes.org">http:/ /www.djernes.org</a>> > 519 Washington Ave. Apt 2, Bridgeville, PA 15017 > > > > > > On Sat, 22 Feb 2003, Ben Clark wrote: > > > >> Does this work? I've tried this same config with no > luck... Each time > >> I call out from asterisk on iconnect it is a > "private number" on the > >> remote caller id. > >> > >> > >> On Saturday, February 22, 2003, at 05:52 PM, Steve > Radich wrote: > >> > >>> I see there's a SetCallerID app I can call; sorry > didn't see that > >>> until > >>> after I sent the mail. > >>> > >>> If anyone else is interested: > >>> > >>> exten => 725,1,SetCallerID,725 > >>> exten => 725,2,Dial,SIP/1(cellnumber)@iconnect > >>> > >>> As has been discussed already - IConnect works > relatively well and > >>> doesn't > >>> tie up a line for the outgoing. If that fails I > can fall through to > >>> a > >>> dial > >>> by Zap, but that's not a concern as voicemail is > always an option. > >>> > >>> Steve Radich - Colocation / Virtual Dedicated / > Dedicated Servers > >>> BitShop, Inc. - <ahref="http://mail.mail1.emailfiltering.co.uk/jump/http://www.bitshop.com">http:/ /www.bitshop.com</a> - $149/month> colo special > >>> > >>> > >>> -----Original Message----- > >>> From: Steve Radich [mailto:stever at bitshop.com] > >>> Sent: Saturday, February 22, 2003 5:58 PM > >>> To: 'asterisk-users at lists.digium.com' > >>> Subject: [Asterisk-Users] Override Caller ID? > >>> > >>> I'm working on a solution for myself to give > different people > >>> different > >>> extensions to reach me; I'm off site quite a bit > and want these > >>> extensions > >>> to fwd to my cell phone when I have call forwarding > on at my desk. > >>> > >>> I want to change the caller id sent to reflect the > extension dialed, > >>> or a > >>> specific caller id - NOT the original callers > caller id - i.e. I want > >>> to > >>> config in my extensions.conf a > Dial/.../callerid=123-456 > >>> > >>> Is there a way to do this? > >>> > >>> It looks relatively easy to patch Dial to do it, > however I'm not > >>> sure I > >>> follow where all the caller id stuff is being > stored/retrieved from > >>> and want > >>> to make sure I'm not patching something that can > already be done. > >>> > >>> Thanks, > >>> > >>> Steve Radich - Colocation / Virtual Dedicated / > Dedicated Servers > >>> BitShop, Inc. - <ahref="http://mail.mail1.emailfiltering.co.uk/jump/http://www.bitshop.com">http:/ /www.bitshop.com</a> - $149/month> colo special > >>> > >>> > >>> -----Original Message----- > >>> From: Steve Radich [mailto:stever at bitshop.com] > >>> Sent: Saturday, February 22, 2003 3:13 PM > >>> To: 'asterisk-users at lists.digium.com' > >>> Subject: [Asterisk-Users] Agressive Echo Cancel > Problem.. > >>> > >>> First let me say the new aggressive echo cancel > seems to work > >>> wonders. > >>> > >>> However in testing I tried a transfer and when I > pressed flash on the > >>> phone > >>> the caller experienced a horrible squealing sound > they said. I > >>> transferred > >>> to another phone I could reach, hit flash again to > join the calls and > >>> heard > >>> this noise myself on the new line joined in - The > original line I > >>> didn't > >>> hear it (or I may have just failed that section of > the hearing test > >>> <grin>). > >>> > >>> Anyone else experiencing this? > >>> > >>> Other than this transfer issue the new echo cancel > sounds great, > >>> > >>> Steve Radich - Colocation / Virtual Dedicated / > Dedicated Servers > >>> BitShop, Inc. - <ahref="http://mail.mail1.emailfiltering.co.uk/jump/http://www.bitshop.com">http:/ /www.bitshop.com</a> - $149/month> colo special > >>> > >>> > >>> -----Original Message----- > >>> From: Klaus-Peter Junghanns > [mailto:kpj at junghanns.net] > >>> Sent: Saturday, February 22, 2003 2:58 PM > >>> To: asterisk-users at lists.digium.com > >>> Subject: Re: [Asterisk-Users] inband DTMF in RTP > >>> > >>> hi mark, > >>> > >>> what's so ugly about this idea? i have modified > chan_sip > >>> to support inband dtmf. it's configurable in > sip.conf on a > >>> per peer basis. > >>> > >>> regards > >>> kapejod > >>> > >>> -- > >>> Klaus-Peter Junghanns > >>> > >>> CEO,CTO > >>> Junghanns.NET Internet-Services & > Software-Development GmbH > >>> Breite Strasse 13 - 12167 Berlin - Germany > >>> fon: +49 30 79705392 > >>> fax: +49 30 79705391 > >>> mobile: +49 160 7503372 > >>> email: kpj at junghanns.net > >>> > >>> > >>> Am Sam, 2003-02-22 um 20.35 schrieb Mark Spencer: > >>>> it could be patched to do so but this is an ugly > idea in general. > >>>> > >>>> Mark > >>>> > >>>> On Thu, 20 Feb 2003, Ben Clark wrote: > >>>> > >>>>> Is it possible to configure asterisk to > understand inband DTMF > >>>>> during > >>> SIP calls? > >>>>> > >>>>> > >>>>> --------------------------------- > >>>>> Do you Yahoo!? > >>>>> Yahoo! Tax Center - forms, calculators, tips, and > more > >>>> > >>>> _______________________________________________ > >>>> Asterisk-Users mailing list > >>>> Asterisk-Users at lists.digium.com > >>>> > <ahref="http://mail.mail1.emailfiltering.co.uk/jump/http://lists.digium.com/mailma n/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk- users</a>> >>> > >>> > >>> _______________________________________________ > >>> Asterisk-Users mailing list > >>> Asterisk-Users at lists.digium.com > >>> > <ahref="http://mail.mail1.emailfiltering.co.uk/jump/http://lists.digium.com/mailma n/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk- users</a>> >>> _______________________________________________ > >>> Asterisk-Users mailing list > >>> Asterisk-Users at lists.digium.com > >>> > <ahref="http://mail.mail1.emailfiltering.co.uk/jump/http://lists.digium.com/mailma n/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk- users</a>> >>> _______________________________________________ > >>> Asterisk-Users mailing list > >>> Asterisk-Users at lists.digium.com > >>> > <ahref="http://mail.mail1.emailfiltering.co.uk/jump/http://lists.digium.com/mailma n/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk- users</a>> >>> _______________________________________________ > >>> Asterisk-Users mailing list > >>> Asterisk-Users at lists.digium.com > >>> > <ahref="http://mail.mail1.emailfiltering.co.uk/jump/http://lists.digium.com/mailma n/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk- users</a>> >>> > >> > >> _______________________________________________ > >> Asterisk-Users mailing list > >> Asterisk-Users at lists.digium.com > >> > <ahref="http://mail.mail1.emailfiltering.co.uk/jump/http://lists.digium.com/mailma n/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk- users</a>> >> > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users at lists.digium.com > > > <ahref="http://mail.mail1.emailfiltering.co.uk/jump/http://lists.digium.com/mailma n/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk- users</a>> > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > <ahref="http://mail.mail1.emailfiltering.co.uk/jump/http://lists.digium.com/mailma n/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk- users</a>