date:20200323

Re: [asterisk-users] Attempting to get BLF working with linphone

2020-03-23 Thread John Hughes


On 23/03/2020 18:51, Joshua C. Colp wrote:
On Mon, Mar 23, 2020 at 2:45 PM John Hughes > wrote:




Why is asterisk giving an error 500? I can find no reason, there
is nothing in any log.


The sequence number is from the past. The first SUBSCRIBE is sequence 
number 22 (check the CSeq header). The second is 20. The third is 21. 
It appears as though this is from the past, so it receives a 500.


Why does asterisk think the error 500 is going to be acked?

It doesn't. The message is for something else, it refers to sequence 
number 103.


Ok, thanks, that's clear and obvious.  Now I have to go beat up(*) the 
linphone people.


((*) in the nicest possible way of course).


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Re: [asterisk-users] Attempting to get BLF working with linphone

2020-03-23 Thread Joshua C. Colp

On Mon, Mar 23, 2020 at 2:45 PM John Hughes  wrote:

> So I've got a bit further with my  project to get BLF working between
> asterisk and linphone.
>
> Initially asterisk was rejecting linphone's SUBSCRIBE messages because
> they didn't have an Accept: header. I've fixed that and now the initial
> SUBSCRIBE messages work and I see all my online contacts in green.
>
> But after a few minutes linphone attempts to renew the subscriptions and
> asterisk is not happy at all:
>
>
> <--- SIP read from UDP:10.27.128.3:5060 --->
> SUBSCRIBE sip:jacques@10.27.128.1:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;rport
> From: ;tag=iGH81k5xf
> To: ;tag=as3c7de68c
> CSeq: 22 SUBSCRIBE
> Call-ID: SQOclJgm4O
> Max-Forwards: 70
> Supported: replaces, outbound
> Event: presence
> Expires: 600
> Accept: application/pidf+xml
> Contact: 
> ;+sip.instance=""
> User-Agent: Linphone/3.12.0 (belle-sip/1.6.3)
> Authorization: Digest realm="asterisk", nonce="188b095b", algorithm=MD5,
> username="john", uri="sip:jacques@10.27.128.1:5060",
> response="bdbc7cbac4453fd643050bf28996a68e"
>
> <->
> --- (14 headers 0 lines) ---
> Found peer 'john' for 'john' from 10.27.128.3:5060
>
> <--- Transmitting (no NAT) to 10.27.128.3:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 10.27.128.3:5060
> ;branch=z9hG4bK.NYP-ux0Zx;received=10.27.128.3;rport=5060
> From: ;tag=iGH81k5xf
> To: ;tag=as3c7de68c
> Call-ID: SQOclJgm4O
> CSeq: 22 SUBSCRIBE
> Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
> nonce="3144c0a9", stale=true
> Content-Length: 0
>
>
> <>
> Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method:
> SUBSCRIBE)
>
> <--- SIP read from UDP:10.27.128.3:5060 --->
> SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0
> Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;rport
> From: ;tag=c3Wvuu2XH
> To: sip:jacq...@masked.masked.com
> CSeq: 20 SUBSCRIBE
> Call-ID: SQOclJgm4O
> Max-Forwards: 70
> Supported: replaces, outbound
> Event: presence
> Expires: 600
> Accept: application/pidf+xml
> Contact: 
> ;+sip.instance=""
> User-Agent: Linphone/3.12.0 (belle-sip/1.6.3)
>
> <->
> --- (13 headers 0 lines) ---
> Sending to 10.27.128.3:5060 (no NAT)
> Creating new subscription
> Sending to 10.27.128.3:5060 (no NAT)
> sip_route_dump: route/path hop: 
> Found peer 'john' for 'john' from 10.27.128.3:5060
>
> <--- Transmitting (no NAT) to 10.27.128.3:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 10.27.128.3:5060
> ;branch=z9hG4bK.oxfLJBaRw;received=10.27.128.3;rport=5060
> From: ;tag=c3Wvuu2XH
> To: sip:jacq...@masked.masked.com;tag=as007ffc64
> Call-ID: SQOclJgm4O
> CSeq: 20 SUBSCRIBE
> Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4224acfb"
> Content-Length: 0
>
>
> <>
> Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method:
> SUBSCRIBE)
>
> <--- SIP read from UDP:10.27.128.3:5060 --->
> SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0
> Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;rport
> From: ;tag=c3Wvuu2XH
> To: sip:jacq...@masked.masked.com
> CSeq: 21 SUBSCRIBE
> Call-ID: SQOclJgm4O
> Max-Forwards: 70
> Supported: replaces, outbound
> Event: presence
> Expires: 600
> Accept: application/pidf+xml
> Contact: 
> ;+sip.instance=""
> User-Agent: Linphone/3.12.0 (belle-sip/1.6.3)
> Authorization: Digest realm="asterisk", nonce="4224acfb", algorithm=MD5,
> username="john", uri="sip:jacq...@masked.masked.com",
> response="eb30a9801e78d2cb2c58c61200c50cb1"
>
> <->
> --- (14 headers 0 lines) ---
>
> <--- Transmitting (no NAT) to 10.27.128.3:5060 --->
> *SIP/2.0 500 Server error*
> Via: SIP/2.0/UDP 10.27.128.3:5060
> ;branch=z9hG4bK.RNv418~xv;received=10.27.128.3;rport=5060
> From: ;tag=c3Wvuu2XH
> To: sip:jacq...@masked.masked.com;tag=as3c7de68c
> Call-ID: SQOclJgm4O
> CSeq: 21 SUBSCRIBE
> Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0
>
>
> <>
>
> [Mar 23 18:23:09] WARNING[2128]: chan_sip.c:4071 retrans_pkt:
> Retransmission timeout reached on transmission SQOclJgm4O for seqno 103
> (Critical Request) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> Packet timed out after 32000ms with no response
>
>
> Why is asterisk giving an error 500? I can find no reason, there is
> nothing in any log.
>

The sequence number is from the past. The first SUBSCRIBE is sequence
number 22 (check the CSeq header). The second is 20. The third is 21. It
appears as though this is from the pa

[asterisk-users] Attempting to get BLF working with linphone

2020-03-23 Thread John Hughes

So I've got a bit further with my  project to get BLF working between 
asterisk and linphone.


Initially asterisk was rejecting linphone's SUBSCRIBE messages because 
they didn't have an Accept: header. I've fixed that and now the initial 
SUBSCRIBE messages work and I see all my online contacts in green.


But after a few minutes linphone attempts to renew the subscriptions and 
asterisk is not happy at all:



<--- SIP read from UDP:10.27.128.3:5060 --->
SUBSCRIBE sip:jacques@10.27.128.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;rport
From: ;tag=iGH81k5xf
To: ;tag=as3c7de68c
CSeq: 22 SUBSCRIBE
Call-ID: SQOclJgm4O
Max-Forwards: 70
Supported: replaces, outbound
Event: presence
Expires: 600
Accept: application/pidf+xml
Contact: 
;+sip.instance=""

User-Agent: Linphone/3.12.0 (belle-sip/1.6.3)
Authorization: Digest realm="asterisk", nonce="188b095b", algorithm=MD5, 
username="john", uri="sip:jacques@10.27.128.1:5060", 
response="bdbc7cbac4453fd643050bf28996a68e"


<->
--- (14 headers 0 lines) ---
Found peer 'john' for 'john' from 10.27.128.3:5060

<--- Transmitting (no NAT) to 10.27.128.3:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;received=10.27.128.3;rport=5060

From: ;tag=iGH81k5xf
To: ;tag=as3c7de68c
Call-ID: SQOclJgm4O
CSeq: 22 SUBSCRIBE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", 
nonce="3144c0a9", stale=true

Content-Length: 0


<>
Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: 
SUBSCRIBE)


<--- SIP read from UDP:10.27.128.3:5060 --->
SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0
Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;rport
From: ;tag=c3Wvuu2XH
To: sip:jacq...@masked.masked.com
CSeq: 20 SUBSCRIBE
Call-ID: SQOclJgm4O
Max-Forwards: 70
Supported: replaces, outbound
Event: presence
Expires: 600
Accept: application/pidf+xml
Contact: 
;+sip.instance=""

User-Agent: Linphone/3.12.0 (belle-sip/1.6.3)

<->
--- (13 headers 0 lines) ---
Sending to 10.27.128.3:5060 (no NAT)
Creating new subscription
Sending to 10.27.128.3:5060 (no NAT)
sip_route_dump: route/path hop: 
Found peer 'john' for 'john' from 10.27.128.3:5060

<--- Transmitting (no NAT) to 10.27.128.3:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;received=10.27.128.3;rport=5060

From: ;tag=c3Wvuu2XH
To: sip:jacq...@masked.masked.com;tag=as007ffc64
Call-ID: SQOclJgm4O
CSeq: 20 SUBSCRIBE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4224acfb"
Content-Length: 0


<>
Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: 
SUBSCRIBE)


<--- SIP read from UDP:10.27.128.3:5060 --->
SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0
Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;rport
From: ;tag=c3Wvuu2XH
To: sip:jacq...@masked.masked.com
CSeq: 21 SUBSCRIBE
Call-ID: SQOclJgm4O
Max-Forwards: 70
Supported: replaces, outbound
Event: presence
Expires: 600
Accept: application/pidf+xml
Contact: 
;+sip.instance=""

User-Agent: Linphone/3.12.0 (belle-sip/1.6.3)
Authorization: Digest realm="asterisk", nonce="4224acfb", algorithm=MD5, 
username="john", uri="sip:jacq...@masked.masked.com", 
response="eb30a9801e78d2cb2c58c61200c50cb1"


<->
--- (14 headers 0 lines) ---

<--- Transmitting (no NAT) to 10.27.128.3:5060 --->
*SIP/2.0 500 Server error*
Via: SIP/2.0/UDP 
10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;received=10.27.128.3;rport=5060

From: ;tag=c3Wvuu2XH
To: sip:jacq...@masked.masked.com;tag=as3c7de68c
Call-ID: SQOclJgm4O
CSeq: 21 SUBSCRIBE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
Content-Length: 0


<>

[Mar 23 18:23:09] WARNING[2128]: chan_sip.c:4071 retrans_pkt: 
Retransmission timeout reached on transmission SQOclJgm4O for seqno 103 
(Critical Request) -- See 
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response


Why is asterisk giving an error 500? I can find no reason, there is 
nothing in any log.


Why does asterisk think the error 500 is going to be acked?


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Re: [asterisk-users] Old Asterisk forums not working

2020-03-23 Thread Joshua C. Colp

On Mon, Mar 23, 2020 at 9:30 AM Jonathan H  wrote:

> Hope you're all well.
>
> I know we should be using  https://community.asterisk.org/ but until
> someone lets Google know that it's moved, all the search results (and
> Asterisk's own search results) come from  https://forums.asterisk.org/
>
> In most browsers, it's not displaying; in Firefox, it says:
>
> "Content Encoding Error
> The page you are trying to view cannot be shown because it uses an invalid
> or unsupported form of compression."
>
> Of course, I can work around by visiting the
> https://community.asterisk.org/  site and searching again there, and it's
> not the most important thing in the world right now, but I thought I'd note
> it anyway.
>

I have raised a ticket with IT regarding this.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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[asterisk-users] Old Asterisk forums not working

2020-03-23 Thread Jonathan H

Hope you're all well.

I know we should be using  https://community.asterisk.org/ but until
someone lets Google know that it's moved, all the search results (and
Asterisk's own search results) come from  https://forums.asterisk.org/

In most browsers, it's not displaying; in Firefox, it says:

"Content Encoding Error
The page you are trying to view cannot be shown because it uses an invalid
or unsupported form of compression."

Of course, I can work around by visiting the https://community.asterisk.org/
site and searching again there, and it's not the most important thing in
the world right now, but I thought I'd note it anyway.

Thanks.
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Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH

2020-03-23 Thread John Hughes


On 23/03/2020 11:29, Joshua C. Colp wrote:
On Mon, Mar 23, 2020 at 7:15 AM John Hughes > wrote:


Hi, in these dark days of COVID-19 lockdown I'm using linphone to
connect to my office asterisk system for working from home.

It's going pretty well but the presence/BLF functions don't appear
to work.

In the linphone logs and asterisk debug I find that asterisk is
rejecting linphone's PUBLISH message:


Asterisk has no support for receiving/storing/using such a PUBLISH 
message. Asterisk instead generates state itself based on whether 
something is on the phone, busy, etc. This is received using a 
SUBSCRIBE and NOTIFY.



Aha!  Thanks a bunch. Now I just have to fix linphone's broken SUBSCRIBE...

Mar 23 11:48:37] WARNING[2128]: chan_sip.c:28198 handle_request_subscribe: 
SUBSCRIBE failure:*no Accept header*: pvt: stateid: -1, laststate: 0, 
dialogver: 0, subscribecont: '', subscribeuri: ''


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Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH

2020-03-23 Thread Social Boh


Because Asterisk do not support PUBLISH.

For BLF Configuration:

https://wiki.asterisk.org/wiki/display/AST/Configuring+chan_sip+for+Presence+Subscriptions

or

https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+for+Presence+Subscriptions

---
I'm SoCIaL, MayBe

On 3/23/20 05:13, John Hughes wrote:
Hi, in these dark days of COVID-19 lockdown I'm using linphone to 
connect to my office asterisk system for working from home.


It's going pretty well but the presence/BLF functions don't appear to 
work.


In the linphone logs and asterisk debug I find that asterisk is 
rejecting linphone's PUBLISH message:


<--- SIP read from UDP:10.27.128.3:5060 --->
PUBLISH sip:j...@xxx.xxx.com SIP/2.0
Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.GRd5yC7Wo;rport
From: ;tag=ZtFgBTxUL
To: sip:j...@xxx.xxx.com
CSeq: 20 PUBLISH
Call-ID: SMHLUSLJD6
Max-Forwards: 70
Supported: replaces, outbound
Event: presence
Accept: application/pidf+xml
Content-Length: 511
Content-Type: application/pidf+xml
Expires: 3600
User-Agent: Linphone/3.12.0 (belle-sip/1.6.3)


xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" 
xmlns:pidfonline="http://www.linphone.org/xsds/pidfonline.xsd"; 
entity="sip:j...@xxx.xxx.com" xmlns="urn:ietf:params:xml:ns:pidf"> 
  open  
 sip:j...@xxx.xxx.com 
2020-03-23T09:40:43Z 


<->
--- (14 headers 3 lines) ---


Sending to 10.27.128.3:5060 (no NAT)

<--- Transmitting (no NAT) to 10.27.128.3:5060 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 
10.27.128.3:5060;branch=z9hG4bK.GRd5yC7Wo;received=10.27.128.3;rport=5060

From: ;tag=ZtFgBTxUL
To: sip:j...@xxx.xxx.com;tag=as674d428f
Call-ID: SMHLUSLJD6
CSeq: 20 PUBLISH
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
Content-Length: 0

I can find nothing in the asterisk logs that says *why* it doesn't 
like the publish.


Help?




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Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH

2020-03-23 Thread Joshua C. Colp

On Mon, Mar 23, 2020 at 7:15 AM John Hughes  wrote:

> Hi, in these dark days of COVID-19 lockdown I'm using linphone to
> connect to my office asterisk system for working from home.
>
> It's going pretty well but the presence/BLF functions don't appear to work.
>
> In the linphone logs and asterisk debug I find that asterisk is
> rejecting linphone's PUBLISH message:
>
> <--- SIP read from UDP:10.27.128.3:5060 --->
> PUBLISH sip:j...@xxx.xxx.com SIP/2.0
> Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.GRd5yC7Wo;rport
> From: ;tag=ZtFgBTxUL
> To: sip:j...@xxx.xxx.com
> CSeq: 20 PUBLISH
> Call-ID: SMHLUSLJD6
> Max-Forwards: 70
> Supported: replaces, outbound
> Event: presence
> Accept: application/pidf+xml
> Content-Length: 511
> Content-Type: application/pidf+xml
> Expires: 3600
> User-Agent: Linphone/3.12.0 (belle-sip/1.6.3)
>
> 
>  xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid"
> xmlns:pidfonline="http://www.linphone.org/xsds/pidfonline.xsd";
> entity="sip:j...@xxx.xxx.com" xmlns="urn:ietf:params:xml:ns:pidf">
>   open 
>  sip:j...@xxx.xxx.com
> 2020-03-23T09:40:43Z 
> 
> <->
> --- (14 headers 3 lines) ---
>
>
> Sending to 10.27.128.3:5060 (no NAT)
>
> <--- Transmitting (no NAT) to 10.27.128.3:5060 --->
> SIP/2.0 489 Bad Event
> Via: SIP/2.0/UDP
> 10.27.128.3:5060;branch=z9hG4bK.GRd5yC7Wo;received=10.27.128.3;rport=5060
> From: ;tag=ZtFgBTxUL
> To: sip:j...@xxx.xxx.com;tag=as674d428f
> Call-ID: SMHLUSLJD6
> CSeq: 20 PUBLISH
> Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0
>
> I can find nothing in the asterisk logs that says *why* it doesn't like
> the publish.
>
> Help?
>

Asterisk has no support for receiving/storing/using such a PUBLISH message.
Asterisk instead generates state itself based on whether something is on
the phone, busy, etc. This is received using a SUBSCRIBE and NOTIFY.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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[asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH

2020-03-23 Thread John Hughes

Hi, in these dark days of COVID-19 lockdown I'm using linphone to 
connect to my office asterisk system for working from home.


It's going pretty well but the presence/BLF functions don't appear to work.

In the linphone logs and asterisk debug I find that asterisk is 
rejecting linphone's PUBLISH message:


<--- SIP read from UDP:10.27.128.3:5060 --->
PUBLISH sip:j...@xxx.xxx.com SIP/2.0
Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.GRd5yC7Wo;rport
From: ;tag=ZtFgBTxUL
To: sip:j...@xxx.xxx.com
CSeq: 20 PUBLISH
Call-ID: SMHLUSLJD6
Max-Forwards: 70
Supported: replaces, outbound
Event: presence
Accept: application/pidf+xml
Content-Length: 511
Content-Type: application/pidf+xml
Expires: 3600
User-Agent: Linphone/3.12.0 (belle-sip/1.6.3)


xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" 
xmlns:pidfonline="http://www.linphone.org/xsds/pidfonline.xsd"; 
entity="sip:j...@xxx.xxx.com" xmlns="urn:ietf:params:xml:ns:pidf"> 
  open  
 sip:j...@xxx.xxx.com 
2020-03-23T09:40:43Z 


<->
--- (14 headers 3 lines) ---


Sending to 10.27.128.3:5060 (no NAT)

<--- Transmitting (no NAT) to 10.27.128.3:5060 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 
10.27.128.3:5060;branch=z9hG4bK.GRd5yC7Wo;received=10.27.128.3;rport=5060

From: ;tag=ZtFgBTxUL
To: sip:j...@xxx.xxx.com;tag=as674d428f
Call-ID: SMHLUSLJD6
CSeq: 20 PUBLISH
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
Content-Length: 0

I can find nothing in the asterisk logs that says *why* it doesn't like 
the publish.


Help?


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Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH

Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH

Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH

[asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH

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