beonice
2005-May-01 18:18 UTC
[Asterisk-Users] Set up SIP, now I'm getting a busy tone and weird (to me) messages on the console
Folks,
I'm hoping someone has already run into this ... the
only other complaint I've seen is here:
http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000640.html
and that basically was a problem with the /etc/hosts
... my server is definitely described in my hosts
file.
I've been using asterisk with IAX and a voicepulse
connect number. No problems at all receiving calls.
Now, I've just purchased a DID in Canada from another
provider, and their proxy only supports SIP. So,
following the generic instructions I've found off the
web, I set up my SIP.conf to point to voicepulse's
server, and set up the other DID to point into this
newly defined sip context, i.e., to
uid:secret@srvr.voicepulse.com/888
The problem? The remote DID, when called, simply gives
me a busy signal. Also, on the asterisk console, I'm
seeing these messages that don't tell me anything:
--------------------------------
May 1 18:37:09 WARNING[12065]: chan_sip.c:695
retrans_pkt: Maximum retries exceeded on call
3d1b58ba507ed7ab2eb141f241b71efb@127.0.0.1 for seqno
103 (Critical Request)
May 1 18:37:23 NOTICE[12065]: chan_sip.c:4036
sip_reg_timeout: -- Registration for
'uid@srvr.voicepulse.com' timed out, trying again
May 1 18:37:23 DEBUG[12065]: chan_sip.c:4150
transmit_register: Scheduled a registration timeout #
5
May 1 18:37:29 WARNING[12065]: chan_sip.c:695
retrans_pkt: Maximum retries exceeded on call
3d1b58ba507ed7ab2eb141f241b71efb@127.0.0.1 for seqno
104 (Critical Request)
May 1 18:37:43 NOTICE[12065]: chan_sip.c:4036
sip_reg_timeout: -- Registration for
'uid@srvr.voicepulse.com' timed out, trying again
May 1 18:37:43 DEBUG[12065]: chan_sip.c:4150
transmit_register: Scheduled a registration timeout #
7
--------------------------------
It looks like the remote DID is failing to register
with the voicepulse server. Any hints on what could be
the problem?
If it helps, here is the relevant portion of my
sip.conf file.
[general]
;context=default ; Default
context for incoming calls
context=unwelcome-calls ; Default context for
incoming calls
; After all, we don't
want any random
; incoming calls to
have access to outbound
; calling - Maya
Kurup, May 1, 2005
;recordhistory=yes ; Record SIP history
by default
; (see sip history /
sip no history)
;realm=mydomain.tld ; Realm for digest
authentication
; defaults to
"asterisk"
; Realms MUST be
globally unique according to RFC 3261
; Set this to your
host name or domain name
port=5060 ; UDP Port to bind to
(SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind
to (0.0.0.0 binds to all)
...
register => uid:secret@srvr.voicepulse.com
; We need to allow at least incoming calls to
; accept calls via libretel, etc.
; So, let's add a context for that:
[888] ; For incoming calls ONLY
type=user ; This device takes incoming calls
username=uid ; Username on device
secret=secret ; Password for device
host=srvr.voicepulse.com ; This host will not
; change frequently
context=allowed_context ; Inbound calls from
; this host go
; to the normal context
-----
and I have allowed_context described in my
extensions.conf, it's the same one I'm using for
regular IAX incoming calls, and works fine.
The context for unwelcome-calls is as follows:
[unwelcome-calls]
;
; Take unknown callers that may have found
; our system, and send them to a re-order tone.
; The string "_." matches any dialed sequence, so all
; calls will result in the Congestion tone application
; being called. They'll get bored and hang up
eventually.
;
exten => _.,1,Congestion
---------------
Any help would be appreciated.
Thanks,
Maya
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Tim Connolly
2005-May-01 18:41 UTC
[Asterisk-Users] Set up SIP, now I'm getting a busy tone and weird (to me)messages on the console
Is NAT=yes on, are you behind a firewall? Give us some connectivity details.
Usually when you see maximum retries, its because you have one-way
communications with the far end for some reason. Are you setting
"externip"
statically?
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of beonice
Sent: Sunday, May 01, 2005 8:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Set up SIP,now I'm getting a busy tone and weird
(to me)messages on the console
Folks,
I'm hoping someone has already run into this ... the
only other complaint I've seen is here:
http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000640.html
and that basically was a problem with the /etc/hosts
... my server is definitely described in my hosts
file.
I've been using asterisk with IAX and a voicepulse
connect number. No problems at all receiving calls.
Now, I've just purchased a DID in Canada from another
provider, and their proxy only supports SIP. So,
following the generic instructions I've found off the
web, I set up my SIP.conf to point to voicepulse's
server, and set up the other DID to point into this
newly defined sip context, i.e., to
uid:secret@srvr.voicepulse.com/888
The problem? The remote DID, when called, simply gives
me a busy signal. Also, on the asterisk console, I'm
seeing these messages that don't tell me anything:
--------------------------------
May 1 18:37:09 WARNING[12065]: chan_sip.c:695
retrans_pkt: Maximum retries exceeded on call
3d1b58ba507ed7ab2eb141f241b71efb@127.0.0.1 for seqno
103 (Critical Request)
May 1 18:37:23 NOTICE[12065]: chan_sip.c:4036
sip_reg_timeout: -- Registration for
'uid@srvr.voicepulse.com' timed out, trying again
May 1 18:37:23 DEBUG[12065]: chan_sip.c:4150
transmit_register: Scheduled a registration timeout #
5
May 1 18:37:29 WARNING[12065]: chan_sip.c:695
retrans_pkt: Maximum retries exceeded on call
3d1b58ba507ed7ab2eb141f241b71efb@127.0.0.1 for seqno
104 (Critical Request)
May 1 18:37:43 NOTICE[12065]: chan_sip.c:4036
sip_reg_timeout: -- Registration for
'uid@srvr.voicepulse.com' timed out, trying again
May 1 18:37:43 DEBUG[12065]: chan_sip.c:4150
transmit_register: Scheduled a registration timeout #
7
--------------------------------
It looks like the remote DID is failing to register
with the voicepulse server. Any hints on what could be
the problem?
If it helps, here is the relevant portion of my
sip.conf file.
[general]
;context=default ; Default
context for incoming calls
context=unwelcome-calls ; Default context for
incoming calls
; After all, we don't
want any random
; incoming calls to
have access to outbound
; calling - Maya
Kurup, May 1, 2005
;recordhistory=yes ; Record SIP history
by default
; (see sip history /
sip no history)
;realm=mydomain.tld ; Realm for digest
authentication
; defaults to
"asterisk"
; Realms MUST be
globally unique according to RFC 3261
; Set this to your
host name or domain name
port=5060 ; UDP Port to bind to
(SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind
to (0.0.0.0 binds to all)
...
register => uid:secret@srvr.voicepulse.com
; We need to allow at least incoming calls to
; accept calls via libretel, etc.
; So, let's add a context for that:
[888] ; For incoming calls ONLY
type=user ; This device takes incoming calls
username=uid ; Username on device
secret=secret ; Password for device
host=srvr.voicepulse.com ; This host will not
; change frequently
context=allowed_context ; Inbound calls from
; this host go
; to the normal context
-----
and I have allowed_context described in my
extensions.conf, it's the same one I'm using for
regular IAX incoming calls, and works fine.
The context for unwelcome-calls is as follows:
[unwelcome-calls]
;
; Take unknown callers that may have found
; our system, and send them to a re-order tone.
; The string "_." matches any dialed sequence, so all
; calls will result in the Congestion tone application
; being called. They'll get bored and hang up
eventually.
;
exten => _.,1,Congestion
---------------
Any help would be appreciated.
Thanks,
Maya
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Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
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