Sven Beisiegel
2006-Dec-05 09:16 UTC
[asterisk-users] No ID from the calling party in SIP Header
Hi... I just started working with Asterisk and found something that looks like an error, but i want to be sure, so that's why I'm asking you. When i make a call from "A" to "B" (both SIP clients), I don't see the name of the called party in the phone that initiated the call, just the dialed number. I made an ethereal trace and found out, that there is no name during the initiation in the SIP Header? But there is a "Remote-Party-ID" in the SIP Packet that goes from the Server to the called party...There is nothing like "P-Asserted-Id" in the SIP Packet that goes to the calling party. My question... Is this an error or did i forget to activate something? The configuration of the sip.conf is: [general] language=de port=5060 disallow=all allow=alaw allow=ulaw allow=GSM nat=no canreinvite=no tos=lowdelay context=default [9001] type=friend username=9001 secret=password host=dynamic callerid=Beckenbauer, Franz <9001> context=default mailbox=9001 callgroup=1 pickupgroup=1 sendrpid=yes [9002] type=friend username=9002 secret=password host=dynamic callerid=Walter, Fritz <9002> context=default mailbox=9002 callgroup=1 pickupgroup=1 sendrpid=yes cheers, Sven
callerid=John Doe <1234> On 05/12/06, Sven Beisiegel <mailsvb@gmail.com> wrote:> > Hi... > > I just started working with Asterisk and found something that looks > like an error, but i want to be sure, so that's why I'm asking you. > > When i make a call from "A" to "B" (both SIP clients), I don't see the > name of the called party in the phone that initiated the call, just > the dialed number. > I made an ethereal trace and found out, that there is no name during > the initiation in the SIP Header? > > But there is a "Remote-Party-ID" in the SIP Packet that goes from the > Server to the called party...There is nothing like "P-Asserted-Id" in > the SIP Packet that goes to the calling party. > > My question... Is this an error or did i forget to activate something? > The configuration of the sip.conf is: > > [general] > language=de > port=5060 > disallow=all > allow=alaw > allow=ulaw > allow=GSM > nat=no > canreinvite=no > tos=lowdelay > context=default > > [9001] > type=friend > username=9001 > secret=password > host=dynamic > callerid=Beckenbauer, Franz <9001> > context=default > mailbox=9001 > callgroup=1 > pickupgroup=1 > sendrpid=yes > > [9002] > type=friend > username=9002 > secret=password > host=dynamic > callerid=Walter, Fritz <9002> > context=default > mailbox=9002 > callgroup=1 > pickupgroup=1 > sendrpid=yes > > > cheers, > Sven > _______________________________________________ > --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/20061208/d818ff7e/attachment.htm