I have an Asterisk server connected to a Nortel Pbx via an E1. Everything works fine, I get calls in and out with callerid. The problem that has been reported to me is the following scenario: A call comes in from the PSTN and is answered by Asterisk. The person dials the operator (1000) which is on the Nortel side so connection is made through the E1. The operator answers and then transfers the call back to a SIP extension on the Asterisk (1303). The result is no audio and a dropped call. My main theory at the moment is that when the receptionist hangs up after the transfer the E1 drops on the Nortel side. Anyone here with this type of integration seen this problem? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120424/23740c3d/attachment.pgp>
Please post your E1 configs. If you are not using QSIG you should. On the nortel side this only works well with R6.0 and later. I have a simular setup but with Cisco UCM but the calls come into the Nortel first and then can be passed back and forth between them with no problem. On 04/24/2012 10:39 AM, Carlos Chavez wrote:> I have an Asterisk server connected to a Nortel Pbx via an E1. > Everything works fine, I get calls in and out with callerid. The > problem that has been reported to me is the following scenario: > > A call comes in from the PSTN and is answered by Asterisk. The person > dials the operator (1000) which is on the Nortel side so connection is > made through the E1. The operator answers and then transfers the call > back to a SIP extension on the Asterisk (1303). The result is no audio > and a dropped call. > > My main theory at the moment is that when the receptionist hangs up > after the transfer the E1 drops on the Nortel side. Anyone here with > this type of integration seen this problem? > > > > -- > _____________________________________________________________________ > -- 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
The E1 between the Asterisk and Nortel is using R2 for signalling. The PSTN comes to Asterisk first and then send calls to the Nortel. When we started we were just replacing an automatic operator/voicemail system for the Nortel and all calls went there. The customer has been gradually shifting extensions to Asterisk and plans to phase out the Nortel completely by next year so we will see this problem crop up more often. On Tue, 2012-04-24 at 11:45 -0500, Jonn Taylor wrote:> Please post your E1 configs. If you are not using QSIG you should. On > the nortel side this only works well with R6.0 and later. I have a > simular setup but with Cisco UCM but the calls come into the Nortel > first and then can be passed back and forth between them with no problem. > > On 04/24/2012 10:39 AM, Carlos Chavez wrote: > > I have an Asterisk server connected to a Nortel Pbx via an E1. > > Everything works fine, I get calls in and out with callerid. The > > problem that has been reported to me is the following scenario: > > > > A call comes in from the PSTN and is answered by Asterisk. The person > > dials the operator (1000) which is on the Nortel side so connection is > > made through the E1. The operator answers and then transfers the call > > back to a SIP extension on the Asterisk (1303). The result is no audio > > and a dropped call. > > > > My main theory at the moment is that when the receptionist hangs up > > after the transfer the E1 drops on the Nortel side. Anyone here with > > this type of integration seen this problem? > > > > > > > > -- > > _____________________________________________________________________ > > -- 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-- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120424/554a2476/attachment.pgp>
You need to change it to QSIG or this will continue to be a problem. On 04/24/2012 12:05 PM, Carlos Chavez wrote:> The E1 between the Asterisk and Nortel is using R2 for signalling. The > PSTN comes to Asterisk first and then send calls to the Nortel. When we > started we were just replacing an automatic operator/voicemail system > for the Nortel and all calls went there. The customer has been > gradually shifting extensions to Asterisk and plans to phase out the > Nortel completely by next year so we will see this problem crop up more > often. > > On Tue, 2012-04-24 at 11:45 -0500, Jonn Taylor wrote: >> Please post your E1 configs. If you are not using QSIG you should. On >> the nortel side this only works well with R6.0 and later. I have a >> simular setup but with Cisco UCM but the calls come into the Nortel >> first and then can be passed back and forth between them with no problem. >> >> On 04/24/2012 10:39 AM, Carlos Chavez wrote: >>> I have an Asterisk server connected to a Nortel Pbx via an E1. >>> Everything works fine, I get calls in and out with callerid. The >>> problem that has been reported to me is the following scenario: >>> >>> A call comes in from the PSTN and is answered by Asterisk. The person >>> dials the operator (1000) which is on the Nortel side so connection is >>> made through the E1. The operator answers and then transfers the call >>> back to a SIP extension on the Asterisk (1303). The result is no audio >>> and a dropped call. >>> >>> My main theory at the moment is that when the receptionist hangs up >>> after the transfer the E1 drops on the Nortel side. Anyone here with >>> this type of integration seen this problem? >>> >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 > > > -- > _____________________________________________________________________ > -- 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
Mc GRATH Ricardo
2012-Apr-24 22:18 UTC
[asterisk-users] Asterisk - Nortel transfer problem
Hi Carlos How about to perform the call transfer operation (screened and unscreened) and trace the communication it will better on Nortel PBX side, in order to know which point is causing call transfer failure. By the way these kind of features is complicated to handler on interconnected system, I just mean is no easy way for check intersystem resource state, as other than inband tones, mainly PBX system interconnect to others system by trunk resource and communication between system are handler by Trunk to Trunk PBX system control operation, PBX only use remote state (Busy, Congestion, etc.) for management own Trunk to Trunk resource operation. Just in case of these kind of scenario it can be done through E1 (DR2 MFCR2 or DTMF) or ISDN PRI QSIG, lastone is more convenience because it cover more supplementary services than DR2 as; Completion of Calls to BusySubscriber (CCBS), Call Hold (HOLD)?by ISDN etc. Best regards Mc GRATH Ricardo E-Mail mcgrathr at mail2web.com
Mc GRATH Ricardo
2012-Apr-25 13:25 UTC
[asterisk-users] Asterisk - Nortel transfer problem
Hi Carlos It could help if you can get a trace of the call transfer from Nortel to SIP extension on the Asterisk (1303), if no way to get from Nortel get from Asterisk. I guest operator try to make a bind call transfer, without wait complete DR2 signalling exchange at least minimal time exchange DR2 signalling between Nortel and Asterisk is about 5 sec. Best regards Mc GRATH Ricardo E-Mail mcgrathr at mail2web.com