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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

head link

[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

head link

[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
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
> 
> --
> _____________________________________________________________________
> -- 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

head link

[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
show
any> > 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

head link

[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

head link

[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

Ian S. Worthington

2011-May-30 18:45 UTC

head link

[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

head link

[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 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: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
> 
> 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

head link

[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 seem
to> > have a way to enter that through the gui), but it didn't fix it:
same
console> > 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

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