Drew Gibson
2007-Mar-28 12:34 UTC
[asterisk-users] Multi-line phones - Asterisk uses wrong callerid
I have some phones (and an ATA) that are shared between two users who each have separate voicemail but they are not behaving as desired nor expected. Incoming calls show up on the correct lines. Calls originating from the device are seen, at the terminating device, as coming from the account listed last in sip.conf, regardless of the line selected. This creates three main issues I would like to resolve:- 1. The person called sees the wrong callerid 2. The CDR records the call against the wrong account 3. Picking up voicemail requires multiple extra steps Is there a way around this?? Scenario:- Phone 1 has three lines 101, 102, 103 Phone 2 has 1 line 202 User 1 selects line 101 at Phone 1 and dials 202 (to Phone 2) User 2 at Phone 2 sees call coming from extension 103 instead of 101 With 'sip debug' enabled at the console, I see an INVITE issued (on the Phone 1 to Asterisk leg) from the correct extension, 101, to 202 but the call leg from Asterisk to Phone 202 shows an INVITE from 103 to 202. 103 happens to be the last listed in sip.conf and the first listed in 'sip show peers' (I have confirmed that this is dependent on the order in the conf file, not numeric order) sip.conf :- [general] port = 5060 bindaddr = 0.0.0.0 pedantic = no autocreatepeer = no context = sip registertimeout=20 localnet = 10.10.10.0/255.255.255.0 srvlookup = yes tos=0xb8 rtptimeout=300 rtpholdtimeout=1800 maxexpirey=3600 defaultexpirey=1200 [sip-101] ; Aastra 480i phones for general office type=peer insecure=very disallow=all allow=ulaw allow=alaw host=dynamic dtmfmode=auto canreinvite=no context=office-dial qualify=yes username=101 secret=xxxxxx mailbox=101 callerid="User 1" <101> sip show peers :- 103/103 10.10.10.181 D 5060 OK (157 ms) 102/102 10.10.10.181 D 5060 OK (159 ms) 202/202 10.10.10.184 D 5060 OK (4 ms) 101/101 10.10.10.181 D 5060 OK (160 ms) Asterisk 1.2.15 Phones tested:- Aastra 480i, Grandstream GXP2000, Grandstream HT-386 ATA -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.com
Matt
2007-Mar-28 13:42 UTC
[asterisk-users] Multi-line phones - Asterisk uses wrong callerid
Do you have multiple devices registering with the 10x extentions? Or is it just the one device? Basically, the phone is not sending the correct Caller-ID, for some reason. Whatever caller-id the phone sends, is what will be sent. On 3/28/07, Drew Gibson <drew@oanda.com> wrote:> > I have some phones (and an ATA) that are shared between two users who > each have separate voicemail but they are not behaving as desired nor > expected. > > Incoming calls show up on the correct lines. > Calls originating from the device are seen, at the terminating device, > as coming from the account listed last in sip.conf, regardless of the > line selected. > > This creates three main issues I would like to resolve:- > 1. The person called sees the wrong callerid > 2. The CDR records the call against the wrong account > 3. Picking up voicemail requires multiple extra steps > > Is there a way around this?? > > Scenario:- > Phone 1 has three lines 101, 102, 103 > Phone 2 has 1 line 202 > > User 1 selects line 101 at Phone 1 and dials 202 (to Phone 2) > User 2 at Phone 2 sees call coming from extension 103 instead of 101 > > With 'sip debug' enabled at the console, I see an INVITE issued (on the > Phone 1 to Asterisk leg) from the correct extension, 101, to 202 but the > call leg from Asterisk to Phone 202 shows an INVITE from 103 to 202. > 103 happens to be the last listed in sip.conf and the first listed in > 'sip show peers' (I have confirmed that this is dependent on the order > in the conf file, not numeric order) > > sip.conf :- > [general] > port = 5060 > bindaddr = 0.0.0.0 > pedantic = no > autocreatepeer = no > context = sip > registertimeout=20 > localnet = 10.10.10.0/255.255.255.0 > srvlookup = yes > tos=0xb8 > rtptimeout=300 > rtpholdtimeout=1800 > maxexpirey=3600 > defaultexpirey=1200 > > [sip-101] > ; Aastra 480i phones for general office > type=peer > insecure=very > disallow=all > allow=ulaw > allow=alaw > host=dynamic > dtmfmode=auto > canreinvite=no > context=office-dial > qualify=yes > username=101 > secret=xxxxxx > mailbox=101 > callerid="User 1" <101> > > > sip show peers :- > 103/103 10.10.10.181 D 5060 OK (157 > ms) > 102/102 10.10.10.181 D 5060 OK (159 > ms) > 202/202 10.10.10.184 D 5060 OK (4 ms) > 101/101 10.10.10.181 D 5060 OK (160 > ms) > > > Asterisk 1.2.15 > Phones tested:- Aastra 480i, Grandstream GXP2000, Grandstream HT-386 ATA > > -- > Drew Gibson > > Systems Administrator > OANDA Corporation > 416-593-6767 x322 > www.oanda.com > > _______________________________________________ > --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/20070328/0c1228f6/attachment.htm
Andrew Joakimsen
2007-Mar-28 21:45 UTC
[asterisk-users] Multi-line phones - Asterisk uses wrong callerid
;------------------------------------------------------------------------------ ; Definitions of locally connected SIP phones ; ; type = user a device that authenticates to us by "from" field to place calls ; type = peer a device we place calls to or that calls us and we match by host ; type = friend two configurations (peer+user) in one ; Thus if you have two "peers" using the same IP address AND port it will probably match. First try to remove insecure=very from your configuration file, that alone might resolve it. If not you need to insure that each line gets its own port. On 3/28/07, Drew Gibson <drew@oanda.com> wrote:> I have some phones (and an ATA) that are shared between two users who > each have separate voicemail but they are not behaving as desired nor > expected. > > Incoming calls show up on the correct lines. > Calls originating from the device are seen, at the terminating device, > as coming from the account listed last in sip.conf, regardless of the > line selected. > > This creates three main issues I would like to resolve:- > 1. The person called sees the wrong callerid > 2. The CDR records the call against the wrong account > 3. Picking up voicemail requires multiple extra steps > > Is there a way around this?? > > Scenario:- > Phone 1 has three lines 101, 102, 103 > Phone 2 has 1 line 202 > > User 1 selects line 101 at Phone 1 and dials 202 (to Phone 2) > User 2 at Phone 2 sees call coming from extension 103 instead of 101 > > With 'sip debug' enabled at the console, I see an INVITE issued (on the > Phone 1 to Asterisk leg) from the correct extension, 101, to 202 but the > call leg from Asterisk to Phone 202 shows an INVITE from 103 to 202. > 103 happens to be the last listed in sip.conf and the first listed in > 'sip show peers' (I have confirmed that this is dependent on the order > in the conf file, not numeric order) > > sip.conf :- > [general] > port = 5060 > bindaddr = 0.0.0.0 > pedantic = no > autocreatepeer = no > context = sip > registertimeout=20 > localnet = 10.10.10.0/255.255.255.0 > srvlookup = yes > tos=0xb8 > rtptimeout=300 > rtpholdtimeout=1800 > maxexpirey=3600 > defaultexpirey=1200 > > [sip-101] > ; Aastra 480i phones for general office > type=peer > insecure=very > disallow=all > allow=ulaw > allow=alaw > host=dynamic > dtmfmode=auto > canreinvite=no > context=office-dial > qualify=yes > username=101 > secret=xxxxxx > mailbox=101 > callerid="User 1" <101> > > > sip show peers :- > 103/103 10.10.10.181 D 5060 OK (157 ms) > 102/102 10.10.10.181 D 5060 OK (159 ms) > 202/202 10.10.10.184 D 5060 OK (4 ms) > 101/101 10.10.10.181 D 5060 OK (160 ms) > > > Asterisk 1.2.15 > Phones tested:- Aastra 480i, Grandstream GXP2000, Grandstream HT-386 ATA > > -- > Drew Gibson > > Systems Administrator > OANDA Corporation > 416-593-6767 x322 > www.oanda.com > > _______________________________________________ > --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 >
Drew Gibson
2007-Mar-29 10:17 UTC
[asterisk-users] Multi-line phones - Asterisk uses wrong callerid
Steve Langstaff wrote:>> -----Original Message----- >> From: asterisk-users-bounces@lists.digium.com >> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >> Drew Gibson >> Sent: 29 March 2007 17:13 >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] Multi-line phones - Asterisk >> uses wrong callerid >> >> Thanks Andrew, I understand the issue now. >> >> Removing "insecure=very" allows the Grandstream phones to >> work, they register separate lines on separate ports (eg Line >> 1=5060, Line 2=5062, etc). >> >> Unfortunately I cannot find a port setting for the Aastra >> 480i, I shall get on their case. >> > > Do you perceive it as a fault with Aastra, for not registering using > separate ports, or Asterisk, for not handling multiple registrations > from the same port? >At this point, I would not like to point fingers, I have not talked to Aastra. 1. Asterisk seems to be assuming that IP/port is the unique key and the account name is less important. 2. Aastra seems to be assuming that account name is the unique key and the IP/port is less important My personal leaning is that account name should be the key (as domain name presides over IP address when web-hosting) but I am not familiar with how this is normally done in the SIP world. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070329/1cdb72c9/attachment.htm