Mark Raming
2004-Nov-07 23:24 UTC
[Asterisk-Users] Problem with call originating from Cisco
> On Sun, 2004-11-07 at 12:06, Mark Raming wrote: > [snip] > > > > ; Incomming calls from Pilmo > > [212.26.192.155] > > type=user > > insecure=yes > > context=incomming > > It is usually spelled as "incoming". Typo or intentional?Yes, this is a typo. But one I made consistently throughout. More like brain malfunction...> > > ; Outgoing calls to Pilmo > > [pilmo] > > type=peer > > insecure=yes > > fromdomain=nelson.ritstele.com > > host=nelson.ritstele.com > > fromuser=31165570909 > > allow=gsm > > disallow=ulaw; > > disallow=alaw; > > dtmfmode=rfc2833 > > The order is: > 1) disallow=all > 2) allow=... > 3) allow=... etc. > > Remove: > allow=gsm > disallow=ulaw; <-- please note the ";" should not be there afaik > disallow=alaw; <-- please note the ";" should not be there afaik > > Try with: > disallow=all > allow=gsmThat's what I currently have: disallow=all, allow=gsm, and that is working but causes Asterisk to do the encoding. What I want is outgoing alaw. So if I remove all disallow/allow lines from the [pilmo] section, asterisk does outgoing alaw (ie it doesn't have to do any transcoding). But then I other party cannot hear me. Thanks, Mark> > Regards, > Patrick > > > > ------------------------------ > > Message: 10 > Date: Sun, 7 Nov 2004 05:45:35 -0800 (PST) > From: jafar mohammed <sonztechnology@yahoo.com> > Subject: [Asterisk-Users] Siemens GSM terminal with Wildcard FXO > To: asterisk-users@lists.digium.com > Message-ID: <20041107134535.96323.qmail@web53706.mail.yahoo.com> > Content-Type: text/plain; charset=us-ascii > > Hi, > > I would like to implement GSM origination for a VOIP > system i am developing. I am purchasing a Siemens M20 > Terminal and would like to know if i can plug it into > my Wildcard FXO device to get incoming GSM calls > routed to the Asterisk server. If anyone has been able > or successful in using this terminal please let me > know. And if any of you have this terminal can you > hook it up to a telephone headset and see if incoming > calls will ring the headset. > > Thank you. > > > > > > __________________________________ > Do you Yahoo!? > Check out the new Yahoo! Front Page. > www.yahoo.com > > > > > ------------------------------ > > Message: 11 > Date: Sun, 7 Nov 2004 09:01:57 -0500 > From: "Steve Totaro" <asterisk@totarotechnologies.com> > Subject: Re: [Asterisk-Users] press # to execute > To: "Mike Roberts" <manipura@gmail.com>, "Asterisk Users > Mailing List > - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <002601c4c4d2$58602a10$0302a8c0@600m> > Content-Type: text/plain; format=flowed; charset="iso-8859-1"; > reply-type=original > > > That would be implimented on the phone. > > Grandstream is like that but on the snom you press OK. > > > ----- Original Message ----- > From: "Mike Roberts" <manipura@gmail.com> > To: <asterisk-users@lists.digium.com> > Sent: Sunday, November 07, 2004 7:08 AM > Subject: [Asterisk-Users] press # to execute > > > >I have this. > > > > exten => 8,1,ANSWER > > exten => 8,2,DigitTimeout,5 > > exten => 8,3,ResponseTimeout,10 > > exten => 8,4,playback(IVR/en_enter_destination) > > > > exten => _1XXXXXXX.,1,dial(SIP/${EXTEN}@146.82.15.241) > > > > Basicaly its like pressing 8 for long distance, but more controled. > > But it has to wait until the timeout before it starts to > dial. Is there > > a way to make them press # when they are done dialing the num > > in order to execute the _1XXXXXXX. I want to turn the timeout up > > but don't want to have them waiting forever. I also need to have a > > "exten => _011." in there as well. So it won't have the same > > amount of digits everytime. > > _______________________________________________ > > 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 > > > > > ------------------------------ > > Message: 12 > Date: Sun, 7 Nov 2004 06:09:51 -0800 > From: Mike Roberts <manipura@gmail.com> > Subject: Re: [Asterisk-Users] press # to execute > To: Steve Totaro <asterisk@totarotechnologies.com> > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <6e0a8bf404110706097e378616@mail.gmail.com> > Content-Type: text/plain; charset=US-ASCII > > I'm trying to do this from PSTN -> DID -> * > > And yes, please spare me the lecture of security, I already know. > > > On Sun, 7 Nov 2004 09:01:57 -0500, Steve Totaro > <asterisk@totarotechnologies.com> wrote: > > > > That would be implimented on the phone. > > > > Grandstream is like that but on the snom you press OK. > > > > > > > > > > ----- Original Message ----- > > From: "Mike Roberts" <manipura@gmail.com> > > To: <asterisk-users@lists.digium.com> > > Sent: Sunday, November 07, 2004 7:08 AM > > Subject: [Asterisk-Users] press # to execute > > > > >I have this. > > > > > > exten => 8,1,ANSWER > > > exten => 8,2,DigitTimeout,5 > > > exten => 8,3,ResponseTimeout,10 > > > exten => 8,4,playback(IVR/en_enter_destination) > > > > > > exten => _1XXXXXXX.,1,dial(SIP/${EXTEN}@146.82.15.241) > > > > > > Basicaly its like pressing 8 for long distance, but more > controled. > > > But it has to wait until the timeout before it starts to > dial. Is there > > > a way to make them press # when they are done dialing the num > > > in order to execute the _1XXXXXXX. I want to turn the timeout up > > > but don't want to have them waiting forever. I also need to have a > > > "exten => _011." in there as well. So it won't have the same > > > amount of digits everytime. > > > _______________________________________________ > > > 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 > > > > > > > > ------------------------------ > > Message: 13 > Date: Sun, 7 Nov 2004 07:04:31 -0800 > From: Mike Roberts <manipura@gmail.com> > Subject: Re: [Asterisk-Users] press # to execute > To: Steve Totaro <asterisk@totarotechnologies.com> > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <6e0a8bf40411070704cee38ce@mail.gmail.com> > Content-Type: text/plain; charset=US-ASCII > > I found it, read() does exactly what I need > > > On Sun, 7 Nov 2004 06:09:51 -0800, Mike Roberts > <manipura@gmail.com> wrote: > > I'm trying to do this from PSTN -> DID -> * > > > > And yes, please spare me the lecture of security, I already know. > > > > > > > > > > On Sun, 7 Nov 2004 09:01:57 -0500, Steve Totaro > > <asterisk@totarotechnologies.com> wrote: > > > > > > That would be implimented on the phone. > > > > > > Grandstream is like that but on the snom you press OK. > > > > > > > > > > > > > > > ----- Original Message ----- > > > From: "Mike Roberts" <manipura@gmail.com> > > > To: <asterisk-users@lists.digium.com> > > > Sent: Sunday, November 07, 2004 7:08 AM > > > Subject: [Asterisk-Users] press # to execute > > > > > > >I have this. > > > > > > > > exten => 8,1,ANSWER > > > > exten => 8,2,DigitTimeout,5 > > > > exten => 8,3,ResponseTimeout,10 > > > > exten => 8,4,playback(IVR/en_enter_destination) > > > > > > > > exten => _1XXXXXXX.,1,dial(SIP/${EXTEN}@146.82.15.241) > > > > > > > > Basicaly its like pressing 8 for long distance, but > more controled. > > > > But it has to wait until the timeout before it starts > to dial. Is there > > > > a way to make them press # when they are done dialing the num > > > > in order to execute the _1XXXXXXX. I want to turn the timeout up > > > > but don't want to have them waiting forever. I also > need to have a > > > > "exten => _011." in there as well. So it won't have the same > > > > amount of digits everytime. > > > > _______________________________________________ > > > > 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 > > > > > > > > > > > > ------------------------------ > > Message: 14 > Date: Sun, 7 Nov 2004 09:36:31 -0600 > From: "Greg Scasny" <gscasny@golden-tech.com> > Subject: RE: [Asterisk-Users] Channel Banks > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > > <12AC7004F95DBA42889BCA2B64A23F68343256@highlife.golden-tech.com> > Content-Type: text/plain; charset="us-ascii" > > Do not buy an adtran - no auto impedance match on FXO ports > (echo, echo, > echo) and no call disconnect supervision on FXO ports. > > Get the ADIT 600 (CAC) (also called a Cactus Lite), they have > all those > features and have a 48 port capability, plus callerid works wonderful. > > If you still want an adtran, I have 3 I can sell you for cheap :) > > Greg > > Gregory P. Scasny > > Golden Technologies Inc. > > http://www.golden-tech.com > > 219-462-7200 > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jay > Brussels > Sent: Friday, November 05, 2004 1:11 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Channel Banks > > We have an Asterisk Server with 5 X100's and a 4-port openline card we > have been using for 6 months. > > My only complaint is echo. Some lines sound good, others echo all the > time, some echo intermittantly or only when conferencing. I > have tweeked the tx/rx gains and played with echo timing as much as > possible. It is time to go to a channel bank (a PRI is still > about $200/month more than POT's lines) . > > It appears the favorite channel banks are CAC and adtran. Am > I missing > anyone? Does one make or model perform better echo > cancallation then the others? Are some channel banks still having > problems with caller-id? > > Jay > > > > > > > > _______________________________________________ > 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 > > > > > ------------------------------ > > Message: 15 > Date: Sun, 7 Nov 2004 17:20:12 +0100 > From: "Nicklas Bondesson" <nicklas.bondesson@mindping.com> > Subject: [Asterisk-Users] No busy-tone > To: <asterisk-users@lists.digium.com> > Message-ID: <20041107162012.D97182FDBF3@lists.digium.com> > Content-Type: text/plain; charset="us-ascii" > > Hi, > > I don't hear a busy-tone when calling an external extension > that's busy. I > just get the Busy Here 486 message in the debugging log. Any ideas? > > Cheers, > Nicklas > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/2 > 0041107/fd484a78/attachment.html > > ------------------------------ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest, Vol 4, Issue 92 > ********************************************* >