Asterisk 13.8 not register with cisco phone 7911g

1

can someone tell how to register cisco phone 7911g with pjsip channels it just running registering
do i need a patch ciscocallmanager please thanks in advances …

2

You most likely need to set “force_rport=no” on the endpoint. Cisco phones don’t adhere to the method defined in the RFC for sending SIP responses, they require that responses go to the address information in the Via header. If the responses go to the source… they are dropped.

3

here is my endpoints

endpoint-internal-d70
type = endpoint
transport = transport-udp-nat
context = Long-Distance
allow = all
direct_media = no
trust_id_outbound = yes
device_state_busy_at = 1
dtmf_mode = rfc4733

auth-userpass
type = auth
auth_type = userpass

aor-single-reg
type = aor
max_contacts = 1

1107
auth = 1107
aors = 1107
callerid = Lindsey Freddie <1107>
force_rport=no i set there am i right please advice

1107
password = 4webrEtHupHewu4
username = 1107

1107
mailboxes = 1107@example

4

You will need to use the preformatted text option to ensure that your configs are not altered. As it is I don’t know.

5

i use xml file like these
i changed the force_rport=no it still says registering so before it worinking fine with sip.conf

here is my xml file:

<?xml version="1.0" encoding="UTF-8"?> true SIP D/M/Y -240 192.168.1.224 directedbroadcast 5060 192.168.1.224 120 true true x-cisco-serviceuri-cfwdall x-cisco-serviceuri-pickup x-cisco-serviceuri-opickup x-cisco-serviceuri-gpickup x-cisco-serviceuri-meetme x-cisco-serviceuri-abbrdial false 1 true true 0 0 0 true false 6 10 180 3600 5 120 120 5 500 4000 70 true Phone 1 false true false false none 101 3 avt false false 0 PBX-ASTERISK 0 true 15000 10 false 16384 32766 false false 9 1107 USECALLMANAGER 5060 1107 1107 1 MODE 1 1107 4webrEtHupHewu4 false 3 0 *99 4 5 true true false true 5 4 2 Speed Dial 44104 5060 184 0 dialtemplate.xml softkeys.xml 1 featurepolicy.xml true 2 false false true 1 0 1 0 1 1 0 1 0 0 1 1 1 0,1,2 1 1 1 0,1 0 1 1 0 0 0 0 0 0 0 1 0 0 0 1 2 2 1 1 1,7 08:00 10:00 00:10 1 1,7 08:00 10:00 00:10 1 0 100 50 5 0 1 0 1 0 d902ed5a-c1e5-4233-b1d6-a960d53d1c3a SIP11.9-4-2SR1-1S 2 Missed Calls Application:Cisco/MissedCalls Received Calls Application:Cisco/ReceivedCalls Placed Calls Application:Cisco/PlacedCalls Voicemail Application:Cisco/Voicemail 1 1 0 http://192.168.1.15/CGI/Execute.pl 1 1 true 0 0 1 5 1 0 0 0 false
6

That’s not useful. You need to use the preformatted text and provide the new pjsip.conf or use a pastebin and provide a link.

7

can you please suggest me a preformatted text how to do that i am new in pjsip please jcold i really appreciated that thanks in advanced

8

It’s an option in this forum. You can do it by using Ctrl+Shift+C. The other option I mentioned, a pastebin, you can do by going to http://pastebin.net/ and pasting in your pjsip.conf and providing the link here.

9

meaning i have to paste my pjsip.conf in the pastebin am i right… or vice versa

10

You paste it in the pastebin and then provide the link so we can look at it.

11

here is the link : http://pastebin.com/iWuPCec2

12

The configuration is correct. Can you pastebin the output of “pjsip set logger on” when the Cisco is attempting to register?

13

here is my xml file cisco: link http://pastebin.com/2JCfMrCc

14

Here is the pjsip debug : http://pastebin.com/zVD5XFve

15

There is no traffic from the Cisco in that log, so it’s likely some sort of configuration issue with the Cisco. I’m unfamiliar with the XML and am unable to help with that.

16

what suggest me before i setup asterisk 11 with ciscocallmanager patch and working putty awsome but with this version i dont if there is patch for asterisk 13.8

17

The traffic is not getting to the channel driver by the looks of it, that’s not something you can patch in PJSIP. It’s likely phone configuration somehow.

18

i solved this issue it was my xml configuration but i have this issue when i tried to call one side and give me this error and i can not hear well please help about that res_hep.c:418 hep_queue_cb: Unable to send packet: Address Family mismatch between source/destination

19

jcold i have an error message like that can you helping me

@186.149.180.225) - No matching endpoint found
[Apr 7 20:53:23] NOTICE[15000]: res_pjsip/pjsip_distributor.c:368 log_unidentified_request: Request from ‘“2130” sip:2130@186.149.180.225’ failed for ‘195.154.181.18:5170’ (callid: 5a950f-592c66-5706f2ef@186.149.180.225) - No matching endpoint found

20

It means there is no endpoint matching the request. In this case presumably no 2130 endpoint.