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" <sip:2062524320 at 89.89.89.253 >;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
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" <sip:2062524320 at 89.89.89.253 > >;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...<snip> 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. 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 jsullivan at opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society
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" <sip:2062524320 at 89.89.89.253 >> ;tag=1c492497235This 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