Good day. Asterisk 13.7.2, res_pjsip. There is a problem of loss of registration of several devices. This happens not on all devices, but problem devices a lot. Below is the log of registration of a contact of one device. Is suspect two things: 1. delete a contact after the contact is added. But, like, it's a feature of code that may already be fixed. 2. deleting a contact much earlier than the 90 seconds specified during the registration Would be grateful for any clues. Dmitriy Serov. expiration settings: [common-aor](!) type=aor qualify_frequency=60 default_expiration=120 maximum_expiration=600 minimum_expiration=90 log: [2016-03-21 20:39:58] VERBOSE[30251] res_pjsip_registrar.c: Added contact 'sip:17367 at 46.39.229.18:37910' to AOR '17367' with expiration of 90 seconds [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:37910 has been created [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:27143 has been deleted [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:37910 is now Reachable. RTT: 41.882 msec [2016-03-21 20:41:01] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:37910 is now Unreachable. RTT: 0.000 msec [2016-03-21 20:41:06] VERBOSE[3827] res_pjsip_registrar.c: Added contact 'sip:17367 at 46.39.229.18:60105' to AOR '17367' with expiration of 90 seconds [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:60105 has been created [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:37910 has been deleted [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:60105 is now Reachable. RTT: 44.031 msec [2016-03-21 20:42:09] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:60105 is now Unreachable. RTT: 0.000 msec [2016-03-21 20:42:14] VERBOSE[3827] res_pjsip_registrar.c: Added contact 'sip:17367 at 46.39.229.18:52836' to AOR '17367' with expiration of 90 seconds [2016-03-21 20:42:14] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:52836 has been created [2016-03-21 20:42:14] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:60105 has been deleted [2016-03-21 20:42:14] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:52836 is now Reachable. RTT: 40.032 msec
Hello guys, I need some help. I have a client coming who wants to assign 5 different numbers to one virtual employee SIP phone at his desk or softphone (Zoiper). which I can assign for the incoming or outgoing both. but the problem is which I might not understanding enough, that, e.g. when line 1 calls the virtual employee will answer ?hello this is xyz company how can I help you? when line 2 calls the virtual employee will answer ?hello this is abc company how can I help you? So it is important the employee can recognize which line is calling as they cannot say the wrong company name by mistake! please let me know if there is any possible ways. currently I have my freeepbx server which I have installed in a VPS server. so all my ZOIPER extension is registered to the Freepbx server with IAX protocol. and I have another Asterisk server at my local office for using SIP phones. basically my both server are connected with IAX protocol as SIP port are blocked in my country. please help if it's possible. thanks in advance On Mon, Mar 21, 2016 at 11:58 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote:> Good day. > > Asterisk 13.7.2, res_pjsip. > There is a problem of loss of registration of several devices. This > happens not on all devices, but problem devices a lot. > Below is the log of registration of a contact of one device. > > Is suspect two things: > 1. delete a contact after the contact is added. But, like, it's a feature > of code that may already be fixed. > 2. deleting a contact much earlier than the 90 seconds specified during > the registration > > Would be grateful for any clues. > > Dmitriy Serov. > > expiration settings: > [common-aor](!) > type=aor > qualify_frequency=60 > default_expiration=120 > maximum_expiration=600 > minimum_expiration=90 > > log: > [2016-03-21 20:39:58] VERBOSE[30251] res_pjsip_registrar.c: Added contact ' > sip:17367 at 46.39.229.18:37910' to AOR '17367' with expiration of 90 seconds > [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:37910 has been created > [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:27143 has been deleted > [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:37910 is now Reachable. RTT: 41.882 > msec > [2016-03-21 20:41:01] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:37910 is now Unreachable. RTT: > 0.000 msec > [2016-03-21 20:41:06] VERBOSE[3827] res_pjsip_registrar.c: Added contact ' > sip:17367 at 46.39.229.18:60105' to AOR '17367' with expiration of 90 seconds > [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:60105 has been created > [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:37910 has been deleted > [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:60105 is now Reachable. RTT: 44.031 > msec > [2016-03-21 20:42:09] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:60105 is now Unreachable. RTT: > 0.000 msec > [2016-03-21 20:42:14] VERBOSE[3827] res_pjsip_registrar.c: Added contact ' > sip:17367 at 46.39.229.18:52836' to AOR '17367' with expiration of 90 seconds > [2016-03-21 20:42:14] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:52836 has been created > [2016-03-21 20:42:14] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:60105 has been deleted > [2016-03-21 20:42:14] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:52836 is now Reachable. RTT: 40.032 > msec > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160322/ea1838a2/attachment.html>
Many desk phones support multiple simultaneous SIP registrations. You could use BLF buttons for each SIP registration and the operator uses the LEDs as their queue as to which account is ringing. Alternatively the phone's UI may be able to indicate which account is ringing without the need for BLFs. Another option is to re-write the CALLERID(num) or CALLERID(name) to indicate the inbound line (eg prepend a string or number). Hopefully that gives you some food for thought :) Pete> On 22/03/2016, at 8:49 am, somsad khan <ctrlz.network at gmail.com> wrote: > <snip> > I have a client coming who wants to assign 5 different numbers to one virtual employee SIP phone at his desk or softphone (Zoiper). > > <snip> > please let me know if there is any possible ways.-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160322/cc00f830/attachment.html>
Richard Mudgett
2016-Mar-21 20:18 UTC
[asterisk-users] Loss of devices registration (pjsip)
On Mon, Mar 21, 2016 at 2:49 PM, somsad khan <ctrlz.network at gmail.com> wrote:> Hello guys, > > I need some help. > > > I have a client coming who wants to assign 5 different numbers to one > virtual employee SIP phone at his desk or softphone (Zoiper). > > > which I can assign for the incoming or outgoing both. > > > but the problem is which I might not understanding enough, that, > > > > e.g. when line 1 calls the virtual employee will answer ?hello this is xyz > company how can I help you? > > when line 2 calls the virtual employee will answer ?hello this is abc > company how can I help you? > > > > So it is important the employee can recognize which line is calling as > they cannot say the wrong company name by mistake! > > > please let me know if there is any possible ways. > > > currently I have my freeepbx server which I have installed in a VPS > server. so all my ZOIPER extension is registered to the Freepbx server with > IAX protocol. and I have another Asterisk server at my local office for > using SIP phones. basically my both server are connected with IAX protocol > as SIP port are blocked in my country. > > > please help if it's possible. thanks in advance >Please do not hijack threads. Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160321/afcd8155/attachment.html>
On Mon, Mar 21, 2016 at 11:58 AM, Dmitriy Serov <serov.d.p at gmail.com> wrote:> Good day. > > Asterisk 13.7.2, res_pjsip. > There is a problem of loss of registration of several devices. This > happens not on all devices, but problem devices a lot. > Below is the log of registration of a contact of one device. > > Is suspect two things: > 1. delete a contact after the contact is added. But, like, it's a feature > of code that may already be fixed. > 2. deleting a contact much earlier than the 90 seconds specified during > the registration > > Would be grateful for any clues. > > Dmitriy Serov. > > expiration settings: > [common-aor](!) > type=aor > qualify_frequency=60 > default_expiration=120 > maximum_expiration=600 > minimum_expiration=90 > > log: > [2016-03-21 20:39:58] VERBOSE[30251] res_pjsip_registrar.c: Added contact ' > sip:17367 at 46.39.229.18:37910' to AOR '17367' with expiration of 90 seconds >?The client just registered?> [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:37910 has been created >?We added a new contact?> [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:27143 has been deleted >?We deleted the old contact?> [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:37910 is now Reachable. RTT: 41.882 > msec >?We qualified the contact successfully?> [2016-03-21 20:41:01] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:37910 is now Unreachable. RTT: > 0.000 msec >?At the next qualify, we couldn't reach the contact [2016-03-21 20:41:06] VERBOSE[3827] res_pjsip_registrar.c: Added contact '> sip:17367 at 46.39.229.18:60105' to AOR '17367' with expiration of 90 seconds >?The client just registered? ?(again)?> [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:60105 has been created >?We added a new contact? [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:37910 has been deleted ?We deleted the old contact?> [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:60105 is now Reachable. RTT: 44.031 > msec >?We qualified the contact successfully?> [2016-03-21 20:42:09] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:60105 is now Unreachable. RTT: > 0.000 msec >?At the next qualify, we couldn't reach the contact ?This looks like a client that's going to sleep or a firewall that's timing out connections. Asterisk is only deleting the contact on the next successful register because it's replacing it. You need to figure out why the qualify is failing and why the client keeps registering. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160321/688ad288/attachment.html>
A J Stiles
2016-Mar-22 08:55 UTC
[asterisk-users] One phone, many names / Was: Loss of devices registration (pjsip)
On Monday 21 Mar 2016, somsad khan wrote:> Hello guys, > > I need some help. > > I have a client coming who wants to assign 5 different numbers to one > virtual employee SIP phone at his desk or softphone (Zoiper). > > which I can assign for the incoming or outgoing both. > > but the problem is which I might not understanding enough, that, > > e.g. when line 1 calls the virtual employee will answer ?hello this is xyz > company how can I help you? > > when line 2 calls the virtual employee will answer ?hello this is abc > company how can I help you? > > So it is important the employee can recognize which line is calling as they > cannot say the wrong company name by mistake! > > please let me know if there is any possible ways.Dead easy! Done this before, in a very similar situation (agent has to answer with a different name, depending on the number the customer dialled). All you need to do -- as long as the phone you are using is modern enough to support it -- is have in your dialplan, before the Dial() instruction to the agent's phone, an instruction like Set(CALLERID(name)=something) where "something" depends on ${EXTEN}. For example, if the numbers for the virtual companies are 731615, 701289 and 718182, and the extension to ring is 301, you might do [from_pstn] ; 731615 is company ABC exten => 731615,1,NoOp(Call to 731615) exten => 731615,n,Set(CALLERID(name)=Company ABC) exten => 731615,n,Dial(301) exten => 731615,n,HangUp() ; 701289 is company XYZ exten => 701289,1,NoOp(Call to 701289) exten => 701289,n,Set(CALLERID(name)=Company XYZ) exten => 701289,n,Dial(301) exten => 701289,n,HangUp() ; 718182 is company PQR exten => 718182,1,NoOp(Call to 718182) exten => 718182,n,Set(CALLERID(name)=Company PQR) exten => 718182,n,Dial(301) exten => 718182,n,HangUp() For the agent to be able to dial out presenting different caller ID numbers, use prefixes such as 16, 17, 18 to indicate dialling out as different companies; strip out the prefix using ${EXTEN:2} to recover the number by skipping two digits from the beginning, and Set(CALLERID(num)=) as appropriate. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk .
On 21 March 2016 at 20:32, George Joseph <george.joseph at fairview5.com> wrote:> > > On Mon, Mar 21, 2016 at 11:58 AM, Dmitriy Serov <serov.d.p at gmail.com> > wrote: > >> Good day. >> >> Asterisk 13.7.2, res_pjsip. >> There is a problem of loss of registration of several devices. This >> happens not on all devices, but problem devices a lot. >> Below is the log of registration of a contact of one device. >> >> Is suspect two things: >> 1. delete a contact after the contact is added. But, like, it's a feature >> of code that may already be fixed. >> 2. deleting a contact much earlier than the 90 seconds specified during >> the registration >> >> Would be grateful for any clues. >> >> Dmitriy Serov. >> >> expiration settings: >> [common-aor](!) >> type=aor >> qualify_frequency=60 >> default_expiration=120 >> maximum_expiration=600 >> minimum_expiration=90 >> >> log: >> [2016-03-21 20:39:58] VERBOSE[30251] res_pjsip_registrar.c: Added contact >> 'sip:17367 at 46.39.229.18:37910' to AOR '17367' with expiration of 90 >> seconds >> > ?The client just registered? > > >> [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367 at 46.39.229.18:37910 has been created >> > ?We added a new contact? > > >> [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367 at 46.39.229.18:27143 has been deleted >> > ?We deleted the old contact? > > >> [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367 at 46.39.229.18:37910 is now Reachable. RTT: >> 41.882 msec >> > ?We qualified the contact successfully? > > >> [2016-03-21 20:41:01] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367 at 46.39.229.18:37910 is now Unreachable. RTT: >> 0.000 msec >> > ?At the next qualify, we couldn't reach the contact > > [2016-03-21 20:41:06] VERBOSE[3827] res_pjsip_registrar.c: Added contact ' >> sip:17367 at 46.39.229.18:60105' to AOR '17367' with expiration of 90 >> seconds >> > ?The client just registered? > > ?(again)? > >> [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367 at 46.39.229.18:60105 has been created >> > ?We added a new contact? > > [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:37910 has been deleted > ?We deleted the old contact? > > >> [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367 at 46.39.229.18:60105 is now Reachable. RTT: >> 44.031 msec >> > ?We qualified the contact successfully? > > >> [2016-03-21 20:42:09] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367 at 46.39.229.18:60105 is now Unreachable. RTT: >> 0.000 msec >> > ?At the next qualify, we couldn't reach the contact > > ?This looks like a client that's going to sleep or a firewall that's > timing out connections. Asterisk is only deleting the contact on the next > successful register because it's replacing it. You need to figure out why > the qualify is failing and why the client keeps registering. > > > > >Check if the router or firewall has a UDP port timeout option and increase it by a lot (I usually up it to an hour). -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160322/08d8ecfe/attachment.html>
Thanks, George Joseph! Now a lot clearer the reasons for this behavior. It turns out that in the case of devices there are two ways to understand that they are "alive": 1. Registration from device to server 2. qualify from the server to the client And the second way does not seems superfluous. Since calling to the device this way will be used. if qualify doesn't working, then this call will not take place. Most likely the problem is that the device is behind two NAT (from your ISP and your own router). Can you advise how to configure the client in this case? Is it necessary to use a stun (did not seem to help, and it only works in the case of RTP) or proxy? Thanks. 21.03.2016 23:32, George Joseph ?????:> > > On Mon, Mar 21, 2016 at 11:58 AM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Good day. > > Asterisk 13.7.2, res_pjsip. > There is a problem of loss of registration of several devices. > This happens not on all devices, but problem devices a lot. > Below is the log of registration of a contact of one device. > > Is suspect two things: > 1. delete a contact after the contact is added. But, like, it's a > feature of code that may already be fixed. > 2. deleting a contact much earlier than the 90 seconds specified > during the registration > > Would be grateful for any clues. > > Dmitriy Serov. > > expiration settings: > [common-aor](!) > type=aor > qualify_frequency=60 > default_expiration=120 > maximum_expiration=600 > minimum_expiration=90 > > log: > [2016-03-21 20:39:58] VERBOSE[30251] res_pjsip_registrar.c: Added > contact 'sip:17367 at 46.39.229.18:37910 > <http://sip:17367 at 46.39.229.18:37910>' to AOR '17367' with > expiration of 90 seconds > > ?The client just registered? > > [2016-03-21 20:39:58] VERBOSE[28019] > res_pjsip/pjsip_configuration.c: Contact > 17367/sip:17367 at 46.39.229.18:37910 > <http://sip:17367 at 46.39.229.18:37910> has been created > > ?We added a new contact? > > [2016-03-21 20:39:58] VERBOSE[28019] > res_pjsip/pjsip_configuration.c: Contact > 17367/sip:17367 at 46.39.229.18:27143 > <http://sip:17367 at 46.39.229.18:27143> has been deleted > > ?We deleted the old contact? > > [2016-03-21 20:39:58] VERBOSE[28019] > res_pjsip/pjsip_configuration.c: Contact > 17367/sip:17367 at 46.39.229.18:37910 > <http://sip:17367 at 46.39.229.18:37910> is now Reachable. RTT: > 41.882 msec > > ?We qualified the contact successfully? > > [2016-03-21 20:41:01] VERBOSE[28019] > res_pjsip/pjsip_configuration.c: Contact > 17367/sip:17367 at 46.39.229.18:37910 > <http://sip:17367 at 46.39.229.18:37910> is now Unreachable. RTT: > 0.000 msec > > ?At the next qualify, we couldn't reach the contact > > [2016-03-21 20:41:06] VERBOSE[3827] res_pjsip_registrar.c: Added > contact 'sip:17367 at 46.39.229.18:60105 > <http://sip:17367 at 46.39.229.18:60105>' to AOR '17367' with > expiration of 90 seconds > > ?The client just registered? > ?(again)? > > [2016-03-21 20:41:06] VERBOSE[28019] > res_pjsip/pjsip_configuration.c: Contact > 17367/sip:17367 at 46.39.229.18:60105 > <http://sip:17367 at 46.39.229.18:60105> has been created > > ?We added a new contact? > [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367 at 46.39.229.18:37910 > <http://sip:17367 at 46.39.229.18:37910> has been deleted > ?We deleted the old contact? > > [2016-03-21 20:41:06] VERBOSE[28019] > res_pjsip/pjsip_configuration.c: Contact > 17367/sip:17367 at 46.39.229.18:60105 > <http://sip:17367 at 46.39.229.18:60105> is now Reachable. RTT: > 44.031 msec > > ?We qualified the contact successfully? > > [2016-03-21 20:42:09] VERBOSE[28019] > res_pjsip/pjsip_configuration.c: Contact > 17367/sip:17367 at 46.39.229.18:60105 > <http://sip:17367 at 46.39.229.18:60105> is now Unreachable. RTT: > 0.000 msec > > ?At the next qualify, we couldn't reach the contact > > ?This looks like a client that's going to sleep or a firewall that's > timing out connections. Asterisk is only deleting the contact on the > next successful register because it's replacing it. You need to figure > out why the qualify is failing and why the client keeps registering. > > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160322/62be3706/attachment.html>
Apparently Analagous Threads
- does res_pjsip support ZRTP?
- Unexpected termination of the call when pick up (res_pjsip)
- [asterisk 13.9] pjsip: Extensions always lost after short period of time
- res_pjsip endpoint config object's 'identify_by' option needs new value "uri".
- res_pjsip endpoint config object's 'identify_by' option needs new value "uri".