Ian S. Worthington
2011-May-28 20:08 UTC
[asterisk-users] Cisco registration problem with 1.8.3.3
I am having a problem registering my cisco phones which is exactly like that described in http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00 The symptoms are: o 7960 lines show [X] o Outbound calls can be made from the phone, including call pickup of inbound calls, but not to it. o Trace shows REGISTER packets sent from phone but no response from Asterisk Is there any way this regressed code could be picked up in a 1833 build or have I got another problem? i
Ryan Wagoner
2011-May-28 20:43 UTC
[asterisk-users] Cisco registration problem with 1.8.3.3
On Sat, May 28, 2011 at 4:08 PM, Ian S. Worthington <ianworthington at usa.net> wrote:> I am having a problem registering my cisco phones which is exactly like that > described in > > http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html > > except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00 > > The symptoms are: > > o 7960 lines show [X] > o Outbound calls can be made from the phone, including call pickup of inbound > calls, but not to it. > o Trace shows REGISTER packets sent from phone but no response from Asterisk > > Is there any way this regressed code could be picked up in a 1833 build or > have I got another problem?I'm able to register a 7940 against Asterisk 1.8.4.1. You might try out that version as it has the fix for registering Cisco phones. However I thought the bug was introduced in 1.8.4 and not 1.8.3.3. I know in the past when I had issues registering Cisco phones I had to make sure the nat settings matched. If you set nat=yes in the sip.conf you must set nat_enable: 1 in SIPDefault.cnf for the phone. What I noticed was when nat=yes is set in Asterisk it ignores the rport and always sends the reply on the port used for the request. Cisco will ignore this reply and not register. Ryan
Ian S. Worthington
2011-May-28 21:18 UTC
[asterisk-users] Cisco registration problem with 1.8.3.3
I too had heard that 1833 did NOT have the 184 problem, which makes me suspicious that it's not that. I don't think its a NAT problem. Neither a sip trace not tcpdump show any response at all to the incoming REGISTER. The phone is on the local lan. I have nat=no and nat_enable: "0" i ------ Original Message ------ Received: 03:45 PM COT, 05/28/2011 From: Ryan Wagoner <rswagoner at gmail.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3> On Sat, May 28, 2011 at 4:08 PM, Ian S. Worthington > <ianworthington at usa.net> wrote: > > I am having a problem registering my cisco phones which is exactly likethat> > described in > > > > http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html > > > > except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00 > > > > The symptoms are: > > > > o 7960 lines show [X] > > o Outbound calls can be made from the phone, including call pickup ofinbound> > calls, but not to it. > > o Trace shows REGISTER packets sent from phone but no response fromAsterisk> > > > Is there any way this regressed code could be picked up in a 1833 buildor> > have I got another problem? > > I'm able to register a 7940 against Asterisk 1.8.4.1. You might try > out that version as it has the fix for registering Cisco phones. > However I thought the bug was introduced in 1.8.4 and not 1.8.3.3. > > I know in the past when I had issues registering Cisco phones I had to > make sure the nat settings matched. If you set nat=yes in the sip.conf > you must set nat_enable: 1 in SIPDefault.cnf for the phone. What I > noticed was when nat=yes is set in Asterisk it ignores the rport and > always sends the reply on the port used for the request. Cisco will > ignore this reply and not register. > > Ryan > > -- > _____________________________________________________________________ > -- 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
Ian S. Worthington
2011-May-29 18:49 UTC
[asterisk-users] Cisco registration problem with 1.8.3.3
Ryan -- Appreciate your continuing assistance with this. (No internet last 24 hours not helping...) The tcpdump was, iiuc, for all traffic on the interface: tcpdump -vv -i eth0 . . . 23:38:58.135264 IP (tos 0x10, ttl 64, id 1172, offset 0, flags [none], proto: UDP (17), length: 49) SIP000785992E4E.50321 > pbx.tftp: [no cksum] 21 RRQ "RINGLIST.DAT" octet 23:38:58.136449 IP (tos 0x0, ttl 64, id 53909, offset 0, flags [none], proto: UDP (17), length: 234) pbx.50006 > SIP000785992E4E.50321: UDP, length 206 23:38:58.139345 IP (tos 0x10, ttl 64, id 1173, offset 0, flags [none], proto: UDP (17), length: 32) SIP000785992E4E.50321 > pbx.50006: [no cksum] UDP, length 4 23:38:58.143704 IP (tos 0x10, ttl 64, id 1174, offset 0, flags [none], proto: UDP (17), length: 49) SIP000785992E4E.50322 > pbx.tftp: [no cksum] 21 RRQ "dialplan.xml" octet 23:38:58.144860 IP (tos 0x0, ttl 64, id 53910, offset 0, flags [none], proto: UDP (17), length: 136) pbx.56660 > SIP000785992E4E.50322: UDP, length 108 23:38:58.146027 IP (tos 0x10, ttl 64, id 1175, offset 0, flags [none], proto: UDP (17), length: 32) SIP000785992E4E.50322 > pbx.56660: [no cksum] UDP, length 4 23:38:58.275694 IP (tos 0x60, ttl 64, id 1176, offset 0, flags [none], proto: UDP (17), length: 377) SIP000785992E4E.50315 > pbx.sip: [no cksum] SIP, length: 349 REGISTER sip:192.168.1.41 SIP/2.0 Via: SIP/2.0/UDP 19\000\000\021\000\000\000\230%\227\011X*\227\011\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000X*\227\011\250w\233\011files\000\000\000\000\000\000\000\021\000\000\000\006\000\000\000`\000\000\000\000\000\000\000\021\000\000\000\263`\202\000\300\266\237\000\000\000\000\000\021\000\000\000x(\227\011H(\227\011rpc\000\031\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000!\000\000\000/lib/libnss_files.so.2\000\000 \000\000\000A\000\000\000tUX\230Ps\003*@]\276+\002\001\002\000\214\272\377\377\000\000\000\000\300\307\377\377\001\004\000\000\260\271\377\377\000\011\000\000BMT\000COST\000COT\000\000\000\000\000\000\000\000\351\001\000\000\003\000\000\000\003\000\000\000\000\000\000\000`\000\000\000\001\000\000\000\000\000\000\000\002\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000 0x0000: 5245 4749 5354 4552 2073 6970 3a31 3932 0x0010: 2e31 3638 2e31 2e34 3120 5349 502f 322e 0x0020: 300d 0a56 6961 3a20 5349 502f 322e 302f 0x0030: 5544 5020 3139 23:38:58.310351 IP6 (hlim 1, next-header: UDP (17), length: 154) fe80::64dd:afa7:ad3a:d38d.52382 > ff02::c.ssdp: UDP, length 146 23:38:58.375689 IP (tos 0x60, ttl 64, id 1177, offset 0, flags [none], proto: UDP (17), length: 377) SIP000785992E4E.50316 > pbx.sip: [no cksum] SIP, length: 349 REGISTER sip:192.168.1.41 SIP/2.0 Via: SIP/2.0/UDP 19\000\000\021\000\000\000\230%\227\011X*\227\011\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000X*\227\011\250w\233\011files\000\000\000\000\000\000\000\021\000\000\000\006\000\000\000`\000\000\000\000\000\000\000\021\000\000\000\263`\202\000\300\266\237\000\000\000\000\000\021\000\000\000x(\227\011H(\227\011rpc\000\031\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000!\000\000\000/lib/libnss_files.so.2\000\000 \000\000\000A\000\000\000tUX\230Ps\003*@]\276+\002\001\002\000\214\272\377\377\000\000\000\000\300\307\377\377\001\004\000\000\260\271\377\377\000\011\000\000BMT\000COST\000COT\000\000\000\000\000\000\000\000\351\001\000\000\003\000\000\000\003\000\000\000\000\000\000\000`\000\000\000\001\000\000\000\000\000\000\000\002\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000 0x0000: 5245 4749 5354 4552 2073 6970 3a31 3932 0x0010: 2e31 3638 2e31 2e34 3120 5349 502f 322e 0x0020: 300d 0a56 6961 3a20 5349 502f 322e 302f 0x0030: 5544 5020 3139 23:38:59.275636 IP (tos 0x60, ttl 64, id 1178, offset 0, flags [none], proto: UDP (17), length: 377) SIP000785992E4E.50315 > pbx.sip: [no cksum] SIP, length: 349 REGISTER sip:192.168.1.41 SIP/2.0 Via: SIP/2.0/UDP 19\000\000\021\000\000\000\230%\227\011X*\227\011\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000X*\227\011\250w\233\011files\000\000\000\000\000\000\000\021\000\000\000\006\000\000\000`\000\000\000\000\000\000\000\021\000\000\000\263`\202\000\300\266\237\000\000\000\000\000\021\000\000\000x(\227\011H(\227\011rpc\000\031\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000!\000\000\000/lib/libnss_files.so.2\000\000 \000\000\000A\000\000\000tUX\230Ps\003*@]\276+\002\001\002\000\214\272\377\377\000\000\000\000\300\307\377\377\001\004\000\000\260\271\377\377\000\011\000\000BMT\000COST\000COT\000\000\000\000\000\000\000\000\351\001\000\000\003\000\000\000\003\000\000\000\000\000\000\000`\000\000\000\001\000\000\000\000\000\000\000\002\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000 0x0000: 5245 4749 5354 4552 2073 6970 3a31 3932 0x0010: 2e31 3638 2e31 2e34 3120 5349 502f 322e 0x0020: 300d 0a56 6961 3a20 5349 502f 322e 302f 0x0030: 5544 5020 3139 23:38:59.375655 IP (tos 0x60, ttl 64, id 1179, offset 0, flags [none], proto: UDP (17), length: 377) SIP000785992E4E.50316 > pbx.sip: [no cksum] SIP, length: 349 REGISTER sip:192.168.1.41 SIP/2.0 Via: SIP/2.0/UDP 19\000\000\021\000\000\000\230%\227\011X*\227\011\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000X*\227\011\250w\233\011files\000\000\000\000\000\000\000\021\000\000\000\006\000\000\000`\000\000\000\000\000\000\000\021\000\000\000\263`\202\000\300\266\237\000\000\000\000\000\021\000\000\000x(\227\011H(\227\011rpc\000\031\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000!\000\000\000/lib/libnss_files.so.2\000\000 \000\000\000A\000\000\000tUX\230Ps\003*@]\276+\002\001\002\000\214\272\377\377\000\000\000\000\300\307\377\377\001\004\000\000\260\271\377\377\000\011\000\000BMT\000COST\000COT\000\000\000\000\000\000\000\000\351\001\000\000\003\000\000\000\003\000\000\000\000\000\000\000`\000\000\000\001\000\000\000\000\000\000\000\002\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000 0x0000: 5245 4749 5354 4552 2073 6970 3a31 3932 0x0010: 2e31 3638 2e31 2e34 3120 5349 502f 322e 0x0020: 300d 0a56 6961 3a20 5349 502f 322e 302f 0x0030: 5544 5020 3139 23:38:59.597018 arp who-has unknown tell Sandy 23:38:59.856114 IP (tos 0x0, ttl 1, id 31181, offset 0, flags [none], proto: UDP (17), length: 161) Sandy.52384 > 239.255.255.250.ssdp: UDP, length 133 23:39:01.275534 IP (tos 0x60, ttl 64, id 1180, offset 0, flags [none], proto: UDP (17), length: 377) SIP000785992E4E.50315 > pbx.sip: [no cksum] SIP, length: 349 REGISTER sip:192.168.1.41 SIP/2.0 Via: SIP/2.0/UDP 19\000\000\021\000\000\000\230%\227\011X*\227\011\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000X*\227\011\250w\233\011files\000\000\000\000\000\000\000\021\000\000\000\006\000\000\000`\000\000\000\000\000\000\000\021\000\000\000\263`\202\000\300\266\237\000\000\000\000\000\021\000\000\000x(\227\011H(\227\011rpc\000\031\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000!\000\000\000/lib/libnss_files.so.2\000\000 \000\000\000A\000\000\000tUX\230Ps\003*@]\276+\002\001\002\000\214\272\377\377\000\000\000\000\300\307\377\377\001\004\000\000\260\271\377\377\000\011\000\000BMT\000COST\000COT\000\000\000\000\000\000\000\000\351\001\000\000\003\000\000\000\003\000\000\000\000\000\000\000`\000\000\000\001\000\000\000\000\000\000\000\002\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000 0x0000: 5245 4749 5354 4552 2073 6970 3a31 3932 0x0010: 2e31 3638 2e31 2e34 3120 5349 502f 322e 0x0020: 300d 0a56 6961 3a20 5349 502f 322e 302f 0x0030: 5544 5020 3139 23:39:01.302688 arp who-has SIP000785992E4E tell pbx 23:39:01.305092 arp reply SIP000785992E4E is-at 00:07:85:99:2e:4e (oui Unknown) 23:39:01.316835 IP6 (hlim 1, next-header: UDP (17), length: 154) fe80::64dd:afa7:ad3a:d38d.52382 > ff02::c.ssdp: UDP, length 146 23:39:01.375519 IP (tos 0x60, ttl 64, id 1181, offset 0, flags [none], proto: UDP (17), length: 377) SIP000785992E4E.50316 > pbx.sip: [no cksum] SIP, length: 349 REGISTER sip:192.168.1.41 SIP/2.0 Via: SIP/2.0/UDP 19\000\000\021\000\000\000\230%\227\011X*\227\011\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000X*\227\011\250w\233\011files\000\000\000\000\000\000\000\021\000\000\000\006\000\000\000`\000\000\000\000\000\000\000\021\000\000\000\263`\202\000\300\266\237\000\000\000\000\000\021\000\000\000x(\227\011H(\227\011rpc\000\031\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000!\000\000\000/lib/libnss_files.so.2\000\000 \000\000\000A\000\000\000tUX\230Ps\003*@]\276+\002\001\002\000\214\272\377\377\000\000\000\000\300\307\377\377\001\004\000\000\260\271\377\377\000\011\000\000BMT\000COST\000COT\000\000\000\000\000\000\000\000\351\001\000\000\003\000\000\000\003\000\000\000\000\000\000\000`\000\000\000\001\000\000\000\000\000\000\000\002\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000 ... Tried changing, as per your suggestion, to nat=yes and your given settings in both SIPDefault.cnf *and* SIPnn....cnf without change. ian ------ Original Message ------ Received: 09:03 PM COT, 05/28/2011 From: Ryan Wagoner <rswagoner at gmail.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3> On Sat, May 28, 2011 at 5:18 PM, Ian S. Worthington > <ianworthington at usa.net> wrote: > > I too had heard that 1833 did NOT have the 184 problem, which makes me > > suspicious that it's not that. > > > > I don't think its a NAT problem. ?Neither a sip trace not tcpdump showany> > response at all to the incoming REGISTER. > > > > The phone is on the local lan. ?I have nat=no and nat_enable: "0" > > > > You are running tcpdump on the Asterisk server? Are you capturing all > traffic or only certain ports? What firmware are you running on the > phone? I am using PS03-8-12-00. It wouldn't hurt to try with nat > enabled, see below. I setup all my phones this way as it saves having > to reconfigure when users take them home. > > sip.conf > nat=yes > > SIPDefault.cnf > nat_enable: 1 > nat_address: "" > nat_received_processing: 1 > > Ryan > > -- > _____________________________________________________________________ > -- 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
Ian S. Worthington
2011-May-29 19:18 UTC
[asterisk-users] Cisco registration problem with 1.8.3.3
And f/w POS3-07-4-00 i
Ryan Wagoner
2011-May-30 12:30 UTC
[asterisk-users] Cisco registration problem with 1.8.3.3
On Sun, May 29, 2011 at 3:18 PM, Ian S. Worthington <ianworthington at usa.net> wrote:> And f/w POS3-07-4-00That is strange that Asterisk is not sending anything back in response to the register. Have you looked at the Asterisk console or logs to see why it is rejecting the register. You might have to enable debug mode core set debug 5 sip set debug on Also if you want to see debug output on the screen check that the following is uncommented in /etc/asterisk/logger.conf console => notice,warning,error,debug Is it possible for you to try a later firmware version? Although 7.4 looks to be a good version according to others notes. http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx Ryan
Ian S. Worthington
2011-May-30 18:45 UTC
[asterisk-users] Cisco registration problem with 1.8.3.3
Ah-ha! Progress at last. (I'd actually tried debug mode before and wondered why I got no output. Any harm in leaving that console => etc enabled?) Console is showing the following. Looks like it doesn't like the format of the REGISTER message??? <--- SIP read from UDP:192.168.1.114:5060 ---> REGISTER sip:192.168.1.41 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK2e11eaa2 From: <sip:702 at 192.168.1.41;user=phone> To: <sip:702 at 192.168.1.41;user=phone> Call-ID: 00078599-2e4e0002-23aa7a4e-0b32ceef at 192.168.1.114 CSeq: 101 REGISTER User-Agent: CSCO/7 Contact: <sip:702 at 192.168.1.114:5060> Content-Length: 0 Expires: 120 <-------------> [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 0 [ 33]: REGISTER sip:192.168.1.41 SIP/2.0 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK2e11eaa2 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 2 [ 39]: From: <sip:702 at 192.168.1.41;user=phone> [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 3 [ 37]: To: <sip:702 at 192.168.1.41;user=phone> [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 4 [ 58]: Call-ID: 00078599-2e4e0002-23aa7a4e-0b32ceef at 192.168.1.114 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 5 [ 18]: CSeq: 101 REGISTER [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 6 [ 18]: User-Agent: CSCO/7 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 7 [ 37]: Contact: <sip:702 at 192.168.1.114:5060> [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 8 [ 17]: Content-Length: 0 [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7941 parse_request: Header 9 [ 12]: Expires: 120 --- (10 headers 0 lines) --- [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7539 find_call: = Looking for Call ID: 00078599-2e4e0002-23aa7a4e-0b32ceef at 192.168.1.114 (Checking From) --From tag --To-tag [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7543 find_call: REGISTER request has no from tag, dropping callid: 00078599-2e4e0002-23aa7a4e-0b32ceef at 192.168.1.114 from: <sip:702 at 192.168.1.41;user=phone> [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:24110 handle_request_do: Invalid SIP message - rejected , no callid, len 337 ian ... ------ Original Message ------ Received: 07:31 AM COT, 05/30/2011 From: Ryan Wagoner <rswagoner at gmail.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3> On Sun, May 29, 2011 at 3:18 PM, Ian S. Worthington > <ianworthington at usa.net> wrote: > > And f/w POS3-07-4-00 > > That is strange that Asterisk is not sending anything back in response > to the register. Have you looked at the Asterisk console or logs to > see why it is rejecting the register. You might have to enable debug > mode > > core set debug 5 > sip set debug on > > Also if you want to see debug output on the screen check that the > following is uncommented in /etc/asterisk/logger.conf > > console => notice,warning,error,debug > > Is it possible for you to try a later firmware version? Although 7.4 > looks to be a good version according to others notes. > > http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx > > Ryan > > -- > _____________________________________________________________________ > -- 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
Ian S. Worthington
2011-May-30 21:18 UTC
[asterisk-users] Cisco registration problem with 1.8.3.3
Many thanks for that. I tried pedantic=no (adding it directly to the [702] section in sip_additional.conf: I'm using the freepbx frontend and it doesn't seem to have a way to enter that through the gui), but it didn't fix it: same console log. Where might I find a reliable source for f/w 8.12? I'm a bit nervous about that as I read that some people feel 7.5 was the last reliable version, and that once you go to 8.x you can't go back? i ------ Original Message ------ Received: 03:53 PM COT, 05/30/2011 From: Ryan Wagoner <rswagoner at gmail.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3> On Mon, May 30, 2011 at 2:45 PM, Ian S. Worthington > <ianworthington at usa.net> wrote: > > Console is showing the following. Looks like it doesn't like the format ofthe> > REGISTER message??? > > > > <--- SIP read from UDP:192.168.1.114:5060 ---> > > REGISTER sip:192.168.1.41 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK2e11eaa2 > > From: <sip:702 at 192.168.1.41;user=phone> > > To: <sip:702 at 192.168.1.41;user=phone> > > Call-ID: 00078599-2e4e0002-23aa7a4e-0b32ceef at 192.168.1.114 > > CSeq: 101 REGISTER > > User-Agent: CSCO/7 > > Contact: <sip:702 at 192.168.1.114:5060> > > Content-Length: 0 > > Expires: 120 > > > > > [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7539 find_call: = Lookingfor> > Call ID: 00078599-2e4e0002-23aa7a4e-0b32ceef at 192.168.1.114 (CheckingFrom)> > --From tag ?--To-tag > > [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:7543 find_call: REGISTERrequest> > has no from tag, dropping callid: > > 00078599-2e4e0002-23aa7a4e-0b32ceef at 192.168.1.114 from: > > <sip:702 at 192.168.1.41;user=phone> > > [2011-05-30 13:33:13] DEBUG[5362]: chan_sip.c:24110 handle_request_do:Invalid> > SIP message - rejected , no callid, len 337 > > The log states "find_call: REGISTER request has no from tag, dropping > callid". If you look at the From: line, it should end with > ;tag=SOMEVALUE. Looking at sip.conf you could set pedantic=no and the > phone should register. The best solution would be to upgrade the phone > firmware. I know 8.12 works. > > Ryan > > -- > _____________________________________________________________________ > -- 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
Ian S. Worthington
2011-May-31 01:36 UTC
[asterisk-users] Cisco registration problem with 1.8.3.3
Sincere thanks Ryan: all is working at long last. I risked the f/w upgrade path in the end rather then something which will be blown away at the next upgrade and leave me scratch me noggin in confusion. Couldn't have done it without your insight. Thanks again. i ------ Original Message ------ Received: 05:11 PM COT, 05/30/2011 From: Ryan Wagoner <rswagoner at gmail.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Cisco registration problem with 1.8.3.3> On Mon, May 30, 2011 at 5:18 PM, Ian S. Worthington > <ianworthington at usa.net> wrote: > > Many thanks for that. > > > > I tried pedantic=no (adding it directly to the [702] section in > > sip_additional.conf: I'm using the freepbx frontend and it doesn't seemto> > have a way to enter that through the gui), but it didn't fix it: sameconsole> > log. > > The setting is a global setting. With FreePBX you want to add > pedantic=no to /etc/asterisk/sip_general_custom.conf You can verify > from the Asterisk console with sip show settings. You should see > Pedantic SIP support: No under Global Signalling Settings > > Ryan > > -- > _____________________________________________________________________ > -- 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