Hi all, I'm trying to get outgoing fax working from an SPA2102 to a PSTN fax machine via a T.38 enabled trunk.? I've got t38pt_udptl = yes faxdetect=no in my sip.conf file.? The ATA has all of the T.38 options turned on, echo cancellation is off, as well as silence suppression off.? The only configured codec is u711.? When the user tries to send a fax, it gets to the point where it issues a reInvite to start the T.38, then the called side receives a SIP 488 (Not Acceptable Here) Where should I start?? Any pointers would be most welcome. -- Take care and have fun, Mike Diehl. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110214/d0a3ea91/attachment.htm>
On Mon, Feb 14, 2011 at 11:00 PM, Mike Diehl <mdiehl at diehlnet.com> wrote:> > Hi all, > > I'm trying to get outgoing fax working from an SPA2102 to a PSTN fax machine > via a T.38 enabled trunk.? I've got > t38pt_udptl = yes > faxdetect=no > > in my sip.conf file.? The ATA has all of the T.38 options turned on, echo > cancellation is off, as well as silence suppression off.? The only > configured codec is u711. > > When the user tries to send a fax, it gets to the point where it issues a > reInvite to start the T.38, then the called side receives a SIP 488 (Not > Acceptable Here) > > Where should I start?? Any pointers would be most welcome. > ________________________________ > Take care and have fun, > Mike Diehl. > -- >Did you turn on sip debugging? I bet you are sending a DID that your provider is not the RespOrg. Some will not take outbound calls with a tollfree number. I screws up the billing when one toll free calls another, who pays for the minutes? So they block it out and if memory serves me correctly, you would get the same message in Asterisk and the call would not go through. It is a shame because if you have toll free DIDs, you probably want them to show up rather than a toll call. I have seen this with XO, you have to plead your case and push alot of minutes for them to break their rules. If your issue is the same as what I had, you have to set the callerid to a number that has been ported over to that provider.? It is too bad because there are many legit reasons to send a number that does not belong to you.. Call forwarding to your cell phone will not work, you will just see the office calling, not the actual caller. I have used it as a GUID, starting at 0000000001 and increment by one, this was in a call center environment that had many people working in Baltimore, MD, Pakistan.?Bogot?, and the Philippines. So a call would come in on a Spanish speaking DID, it would be sent to Bogot? with the callerid of whatever I sent. At the end of the month, when it was time to settle with the remote call centers, we had a way to really audit the bill rather than just paying it. The destination, GUID, CDRs all stored in a database. We also recorded using Orecx and and tied that into the same database and had full integration with the home brewed CRM. Thanks, Steve Totaro
I am wondering if its possible to have sometime like this: exten 100 => Dial (g/08039269311) where g would be a group of SIP extensions and i would be parsing/hard coding the PSTN numbers into it, so when i dial extension 100, it passes the call to a group of SIP service provider extensions. i would also like to know if that is possible, i am planning to write a line of code that would parse in PSTN numbers asynchronously, would like to know if the groups would be dialed synchronously. Thanks, Best Regards, Abejide Ayodele (CCNA)+2348039269311 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110215/860142e3/attachment.htm>
On 02/14/2011 10:00 PM, Mike Diehl wrote:> Hi all, > > I'm trying to get outgoing fax working from an SPA2102 to a PSTN fax machine > via a T.38 enabled trunk. I've got > t38pt_udptl = yes > faxdetect=no > > in my sip.conf file. The ATA has all of the T.38 options turned on, echo > cancellation is off, as well as silence suppression off. The only > configured codec is u711. > > When the user tries to send a fax, it gets to the point where it issues a > reInvite to start the T.38, then the called side receives a SIP 488 (Not > Acceptable Here) > > Where should I start? Any pointers would be most welcome.As is always the case, posting a complete (*all* logger levels enabled) log capture, with SIP debugging enabled, is the way to get started. Nobody can effectively help you without seeing exactly what is happening first. It would also help if you didn't use the word 'it' to mean different things in a single sentence :-) Clarity and completeness make it much easier for people to understand what you are trying to express. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kfleming at digium.com Check us out at www.digium.com & www.asterisk.org
this command will not work. what is your main purpose? do u need to have a conference with a group of sip phones? best On Tue, Feb 15, 2011 at 3:13 PM, ayodele abejide <ayodeleabejide at hotmail.com> wrote:> I am wondering if its possible to have sometime like this: > > exten 100 => Dial (g/08039269311) > > where g would be a group of SIP extensions and i would be parsing/hard > coding the PSTN numbers into it, so when i dial extension 100, it passes the > call to a group of SIP service provider extensions. > > i would also like to know if that is possible, i am planning to write a > line of code that would parse in PSTN numbers asynchronously, would like to > know if the groups would be dialed synchronously. > > Thanks, > > Best Regards, > > Abejide Ayodele (CCNA) > +2348039269311 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110215/b4ca52bc/attachment.htm>
Hi, I want to trunk outbound calls through a sip provider to PSTN, and i want to write a script to parse the PSTN numbers, so when say extension 100 is dialled it just starts to dial PSTN numbers through the SIP provider, so it justs an automated dialer. Best Regards, ABEJIDE, Ayodele A. (CCNA) +2348039269311 "Before long, paying for a phone call will be as alien as paying for email" From: lopl at lopl.net Date: Tue, 15 Feb 2011 16:14:22 +0330 To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Dial command this command will not work. what is your main purpose? do u need to have a conference with a group of sip phones? best On Tue, Feb 15, 2011 at 3:13 PM, ayodele abejide <ayodeleabejide at hotmail.com> wrote: I am wondering if its possible to have sometime like this: exten 100 => Dial (g/08039269311) where g would be a group of SIP extensions and i would be parsing/hard coding the PSTN numbers into it, so when i dial extension 100, it passes the call to a group of SIP service provider extensions. i would also like to know if that is possible, i am planning to write a line of code that would parse in PSTN numbers asynchronously, would like to know if the groups would be dialed synchronously. Thanks, Best Regards, Abejide Ayodele (CCNA)+2348039269311 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110215/5cb9cf93/attachment.htm>
On Tue, Feb 15, 2011 at 01:06:16PM +0000, ayodele abejide wrote:> I want to trunk outbound calls through a sip provider to PSTN, and i > want to write a script to parse the PSTN numbers, so when say > extension 100 is dialled it just starts to dial PSTN numbers through > the SIP provider, so it justs an automated dialer.It seems your are looking for the AGI command. Where you can write a script to build a multi device dial (eg Dial(SIP/123 at foo&SIP/234 at bar)) depending on the dialed number. Take a look at http://www.voip-info.org/wiki/view/Asterisk+AGI and get an AGI class for your favorite scripting language. -- Daniel Tryba