I am having a strange problem with an asterisk servier using R2 Unicall in Mexico. Most calls go through fine but some of them give me an error like this: -- Executing Dial("SIP/86-db41", "Unicall/g2/014448343600") in new stack -- Called g2/014448343600 Feb 9 21:44:39 WARNING[23069]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Dialing Feb 9 21:44:45 WARNING[23069]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Protocol failure -- Unicall/2 protocol error. Cause 32769 Feb 9 21:44:45 WARNING[23069]: app_dial.c:705 wait_for_answer: Unable to forward voice Feb 9 21:44:45 WARNING[23069]: app_dial.c:705 wait_for_answer: Unable to forward voice -- Hungup 'UniCall/2-1' == Everyone is busy/congested at this time (1:0/0/1) This particular call is national long distance. But I have seen the problem with some local numbers. I even had a problem dialing a company in the same city, their main numbers gave this error but their fax number went through without problem. I am using Asterisk 1.2.4 (upgraded from 1.2.3 this morning), spandsp .21, unicall 0.0.3. Any ideas? I am using a TE110P card with Zaptel 1.2.3 with 10 channels from Telmex. -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001
Try to change the value of protocolvariant in the unicall.conf. Please, send us here the result. Darlon Ferreira Bortolini Rede/Desenvolvimento Betha Sistemas Fone (48) 3431-0750/Ramal 1000 ----- Original Message ----- From: Carlos Chavez To: Asterisk Sent: Friday, February 10, 2006 1:57 AM Subject: [Asterisk-Users] Problem win Unicall I am having a strange problem with an asterisk servier using R2 Unicall in Mexico. Most calls go through fine but some of them give me an error like this: -- Executing Dial("SIP/86-db41", "Unicall/g2/014448343600") in new stack -- Called g2/014448343600 Feb 9 21:44:39 WARNING[23069]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Dialing Feb 9 21:44:45 WARNING[23069]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Protocol failure -- Unicall/2 protocol error. Cause 32769 Feb 9 21:44:45 WARNING[23069]: app_dial.c:705 wait_for_answer: Unable to forward voice Feb 9 21:44:45 WARNING[23069]: app_dial.c:705 wait_for_answer: Unable to forward voice -- Hungup 'UniCall/2-1' == Everyone is busy/congested at this time (1:0/0/1) This particular call is national long distance. But I have seen the problem with some local numbers. I even had a problem dialing a company in the same city, their main numbers gave this error but their fax number went through without problem. I am using Asterisk 1.2.4 (upgraded from 1.2.3 this morning), spandsp .21, unicall 0.0.3. Any ideas? I am using a TE110P card with Zaptel 1.2.3 with 10 channels from Telmex. -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- 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/20060210/4c20400c/attachment.htm
Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060210/32f9473b/attachment.pgp
Hi Carlos , how do you did this part ? " I also included a bit timeout of 120 seconds in the dial command." Thanks in advanced. Regards Athiel 2006/2/10, Carlos Chavez <cursor@telecomabmex.com>:> > On Fri, 2006-02-10 at 08:38 -0200, Darlon wrote: > > Try to change the value of protocolvariant in the unicall.conf. Please, > send us here the result. > > > > > > I am using mx,10,4 in the protocol variant of unicall.conf. What > seemed to solve the problem is a very old tip that said I should change the > DEFAULT_T1 value of mfcr2.c fomr 5000 to something like 20000. I also > included a bit timeout of 120 seconds in the dial command. For the moment > every call is going through although I still have some testing to do. > > > -- > Carlos Chavez > Director de Tecnolog?a > Telecomunicaciones Abiertas de M?xico S.A. de C.V. > Tel: +52-55-91169161 Ext 2001 > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.1 (GNU/Linux) > > iD8DBQBD7OR0Vhw7eWImqUMRAg3HAJ0bzfK7twgfRueZuhp984FQO91EoQCcCcfw > auI71ZV1Cu3ZI+sMNkT52Wk> =3e1I > -----END PGP SIGNATURE----- > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > 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/20060221/630ca377/attachment.htm
Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 191 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060221/6362bd1d/attachment.pgp