Re: [asterisk-users] Question about callerid?
OK, Now I am responding to myself, because I have figured it out (finally). It turns out it's a "feature" of asterisk (at least the older versions). This is where I found my answer: https://issues.asterisk.org/view.php?id=9678 So the solution for me was to simply rearrange my sip.conf so my incoming call handling peer is at the very end. Pretty wacky. I am hopefully back on the road though with working caller ID as well. Marty On Nov 14, 2009, at 11:10 AM, Martin Joseph wrote: > Ok I am replying to myself, because I still don't have this figured > out,, but I think I have more info. > > > On Nov 5, 2009, at 8:57 PM, Martin Joseph wrote: > >> >> Hello again Asterisk people. >> >> I am running Asterisk 1.42 on an old PowerPC ibook. I have had this >> deployed for several years now, with pretty good results. >> >> Recently I added a callerid service to my landline (qwest). >> >> I am using the audiocodes MP114 2fxo/2fxs gateway, which is an >> outstanding piece of hardware once it's configured (lol). > I think the issue is related to the fact that the MP114 is in my case > a combination device. 2fxo/2fxs setup. > > It seems like what happens is when a call comes into the fxo it is > inviting asterisk with the correct callerid information(sip from). > Asterisk attempts to use this invite as a basis for a new call. > > HOWEVER, for some reason or another (bug?) Asterisk identifies the fxs > extension at the same IP address as a peer for the basis of the new > call, and since the other peer (friend) is the FXS, the authentication > fails, and caller ID is lost. > > If I remove my FXS (friend) definition from sip.conf then suddenly all > is well and the the callerID string is passed aok. Of course then > none of the phones attached to the FXS work, which is a problem... > > I hope someone has some ideas on what I am doing wrong/some way to fix > this? > > Thanks in advance for any help you might offer. > > Marty > >> >> Anyhow, I can see that the gateway is passing caller id info to >> asterisk because the console will display something like: >> >> [Nov 4 13:01:19] NOTICE[32]: chan_sip.c:13362 handle_request_invite: >> Failed to authenticate user "SEATTLE SCHOOLS" >> ;tag=1c492497235 > This authentication is failing because of the mismatch of extensions > described above. The FXO is ext. 2003 and the FXS is ext. 2005. >> >> So the caller ID info is right there. >> >> However on my extensions (or softphones) the id shows as the >> extension >> # (ie 2003). >> >> Is there something I need to do to set the callerid? I can't seem to >> find this in the examples? >> >> Thanks in advance for helping with my (I am sure) stupid question... >> >> Marty >> >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> 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 -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about callerid?
Ok I am replying to myself, because I still don't have this figured out,, but I think I have more info. On Nov 5, 2009, at 8:57 PM, Martin Joseph wrote: > > Hello again Asterisk people. > > I am running Asterisk 1.42 on an old PowerPC ibook. I have had this > deployed for several years now, with pretty good results. > > Recently I added a callerid service to my landline (qwest). > > I am using the audiocodes MP114 2fxo/2fxs gateway, which is an > outstanding piece of hardware once it's configured (lol). I think the issue is related to the fact that the MP114 is in my case a combination device. 2fxo/2fxs setup. It seems like what happens is when a call comes into the fxo it is inviting asterisk with the correct callerid information(sip from). Asterisk attempts to use this invite as a basis for a new call. HOWEVER, for some reason or another (bug?) Asterisk identifies the fxs extension at the same IP address as a peer for the basis of the new call, and since the other peer (friend) is the FXS, the authentication fails, and caller ID is lost. If I remove my FXS (friend) definition from sip.conf then suddenly all is well and the the callerID string is passed aok. Of course then none of the phones attached to the FXS work, which is a problem... I hope someone has some ideas on what I am doing wrong/some way to fix this? Thanks in advance for any help you might offer. Marty > > Anyhow, I can see that the gateway is passing caller id info to > asterisk because the console will display something like: > > [Nov 4 13:01:19] NOTICE[32]: chan_sip.c:13362 handle_request_invite: > Failed to authenticate user "SEATTLE SCHOOLS" > ;tag=1c492497235 This authentication is failing because of the mismatch of extensions described above. The FXO is ext. 2003 and the FXS is ext. 2005. > > So the caller ID info is right there. > > However on my extensions (or softphones) the id shows as the extension > # (ie 2003). > > Is there something I need to do to set the callerid? I can't seem to > find this in the examples? > > Thanks in advance for helping with my (I am sure) stupid question... > > Marty > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about callerid?
On Nov 6, 2009, at 5:14 AM, John A. Sullivan III wrote: > On Thu, 2009-11-05 at 20:57 -0800, Martin Joseph wrote: >> Hello again Asterisk people. >> >> I am running Asterisk 1.42 on an old PowerPC ibook. I have had this >> deployed for several years now, with pretty good results. >> >> Recently I added a callerid service to my landline (qwest). >> >> I am using the audiocodes MP114 2fxo/2fxs gateway, which is an >> outstanding piece of hardware once it's configured (lol). >> >> Anyhow, I can see that the gateway is passing caller id info to >> asterisk because the console will display something like: >> >> [Nov 4 13:01:19] NOTICE[32]: chan_sip.c:13362 handle_request_invite: >> Failed to authenticate user "SEATTLE SCHOOLS" >> ;tag=1c492497235 >> >> So the caller ID info is right there. >> >> However on my extensions (or softphones) the id shows as the >> extension >> # (ie 2003). >> >> Is there something I need to do to set the callerid? I can't seem to >> find this in the examples? >> >> Thanks in advance for helping with my (I am sure) stupid question... > > I'd like to understand this better myself as I know we don't have this > right in our environment. I believe the reason you see that is > because > Asterisk is providing a B2BUA (I think it's called), i.e., your caller > is not actually talking to your phone. Instead, your caller is > talking > to Asterisk on the inbound SIP ID (whatever that is) and then Asterisk > is calling your phone from the extension in the dial plan. At least I > think that's why the extension shows up in the callerID. OK, that makes sense. So since Asterisk is a back to to back user agent (ie the call is always going through it) then the Caller ID data isn't magically moved along... Still, the fact that it's showing up there in the console means there should be some way to grab it (the callerID data) and stuff into into the proper place for it to be passed along. I see that the callerid valiable can be set as per: http://www.voip-info.org/wiki/view/Setting+Callerid So that's nice, and the only question is how to I get the callerID info from where it show in the console as "failed to authenticate"? Either that, or I could reconfigure my audiocodes and my asterisk so that instead of incoming calls dialing my desired extension (ie 2020), asterisk could accept the calls from the domain of the audiocodes (ie it's IP address). Maybe that's how get the CID data. Don't really know, but suspect there are lots of people here who do? Thanks for any help in advance, Marty > > The identity can be overridden in sip.conf with the fromdomain and > fromuser parameters. However, we found this introduced its own > problems. I suppose we just need to build more sophisticated logic > into > our dialplan. The problem is, if we set the fromdomain/user, we now > show correct sip sources when we make direct SIP calls and can return > those calls from the phone's call history. However, it breaks all the > internal dialing which wants to dial to the extension. If we remove > fromdomain/user, the internal dialing works but public SIP calls now > show the extension as the user rather than the user's public SIP ID. > > I'm sure as with most things in Asterisk, we can fix it if we just > take > the time to think through the programming logic. Hope this helps - > John > -- > John A. Sullivan III > Open Source Development Corporation > +1 207-985-7880 > jsulli...@opensourcedevel.com > > http://www.spiritualoutreach.com > Making Christianity intelligible to secular society > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about callerid?
Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding piece of hardware once it's configured (lol). Anyhow, I can see that the gateway is passing caller id info to asterisk because the console will display something like: [Nov 4 13:01:19] NOTICE[32]: chan_sip.c:13362 handle_request_invite: Failed to authenticate user "SEATTLE SCHOOLS" ;tag=1c492497235 So the caller ID info is right there. However on my extensions (or softphones) the id shows as the extension # (ie 2003). Is there something I need to do to set the callerid? I can't seem to find this in the examples? Thanks in advance for helping with my (I am sure) stupid question... Marty ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about Callerid
Hi all, i have one question: Why the callerid is not set in the "Newchannel" event if the channel is not a Zap channel? I think the callerid is known at this time, is it not ? The both cases i mean are when i make a call over a H323 softphone or a I4L channel . Can anybody help ?? Thanks, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
