> I am facing an issue with Peer registration in my asterisk server .
>
> I am using asterisk version 1.8.5.0 and using SIP real-time
> architecture.when i am doing registration it registered fine on asterisk
> as peer is available in Database.
>
> But now i am doing ‘sip reload’ or ‘reload’ due to some reason my peer
> registration is going out and i cannot able to call that peer even though
> in SIP client it shows me ‘registered’.
>
> Can any body elaborate on this issue which settings i need to put in
> sip.conf.
>
> I also tried to follow this patch
> https://issues.asterisk.org/view.php?id=14196 But it allready applied in
> code base so why it wont work?
>
> Here is my sip.conf settings.
>
> [general]
> context=from-internal        ; Default context for incoming cal
> rtcachefriends=no
> rtupdate=yes
> rtautoclear=yes
> rtsavesysname=yes
> callcounter = yes
> callevents=yes
> bindport=5060            ; UDP Port to bind to (SIP standard port is 5060)
> srvlookup=yes            ; Enable DNS SRV lookups on outbound calls
> pedantic=yes            ; Enable slow, pedantic checking for Pingtel
> tos=184            ; Set IP QoS to either a keyword or numeric val
> tos_sip=cs3                    ; Sets TOS for SIP packets.
> tos_audio=ef                   ; Sets TOS for RTP audio packets.
> tos=lowdelay            ; lowdelay,throughput,reliability,mincost,none
> maxexpiry=3600            ; Max length of incoming registration we allow
> defaultexpiry=120        ; Default length of incoming/outoing registration
> preferred_codec_only=yes
> disallow=all            ; First disallow all codecs
> allow=ulaw            ; Allow codecs in order of preference
> allow=alaw
> insecure=invite
> language=en                   ; Default language setting for all
> users/peers
> rtpholdtimeout=300        ; Terminate call if 300 seconds of no RTP
> activity
> useragent=dhaval              ; Allows you to change the user agent string
> dtmfmode = rfc2833        ; Set default dtmfmode for sending DTMF. Default:
> rfc2833
> qualify=yes
> nat=yes
> ;canreinvite=yes
> directmedia=yes
> directrtpsetup=yes
>
> And here is DB fields snapshots.
>
>               id: 1
>             name: 201
>           ipaddr: 172.18.100.243
>             port: 53624
>       regseconds: 1328716180
>      defaultuser: 201
>      fullcontact: NULL
>        regserver: dhaval
>        useragent: CSipSimple r1133 / b
>           lastms: 554
>             host: dynamic
>             type: friend
>          context: from-internal
>           permit: NULL
>             deny: NULL
>           secret: 201
>        md5secret: NULL
>     remotesecret: NULL
>        transport: NULL
>         dtmfmode: NULL
>      directmedia: yes
>              nat: NULL
>            allow: ulaw
>         disallow: g729
>         insecure: invite
>         callerid: NULL
> rfc2833compensate: NULL
>          mailbox: NULL
>   session-timers: NULL
>  session-expires: NULL
>    session-minse: NULL
> session-refresher: NULL
>
> Kindly help me to resolve this.
>
> Thanks
> Dhaval
>

The first thing I would try is ‘rtcachefriends=yes’, that should do it.

JR

Hello,

I am trying to deposit a voicemail message(using voicemail() application) for a subscriber using asterisk-1.8.7.1. But i am facing  aproblem in the rtp port allocation for a session due to which ’488 Not Acceptable’ response is sent towards the client end.  Following are error messages:

[Jan 18 12:43:59] ERROR[19164] res_rtp_asterisk.c: Failed to Allocate port 7660 for RTP instance ’0x1a75ab98′
[Jan 18 12:43:59] ERROR[19164] res_rtp_asterisk.c: Oh dear… we couldn’t allocate a port (x=7662)7660 for RTP instance ’0x1a75ab98′. errno 99
[Jan 18 12:43:59] DEBUG[19164] rtp_engine.c: Engine ‘asterisk’ failed to setup RTP instance ’0x1a75ab98′
[Jan 18 12:43:59] DEBUG[19164] rtp_engine.c: Destroyed RTP instance ’0x1a75ab98′
[Jan 18 12:43:59] DEBUG[19164] chan_sip.c: ERROR: failed to allocate rtp instance
[Jan 18 12:43:59] DEBUG[19164] chan_sip.c: Could not initialize RTP instance for dialog: 800E51A5-1140-E111-A216-001A4B4698C3@10.34.77.90

Please  find attached the log file  for more information.

Regards,
Shalu

messages_rtp_port_allocation_problem.zip

Hi all again,

I’ve got a problem with sporadic one way audio calls, which means
sometimes I can’t hear the calling party (call is established, but audio
is missing). Today I received ~90 calls, one of them got this problem.

I’ve got two networks involved, without NAT:

- 192.168.1.X, in there one nic of my server and all the phones
- a private net to my provider, in there a nic of my server and the voice
switch of my provider

My sip.conf:
[general]
port=5060
bindaddr=0.0.0.0
language=de
allowguest=no
;echocancel=yes
;echotraining=yes
alwaysauthreject=yes
disallow=all
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=XXX.XXX.X.X/29
permit=192.168.1.0/24
;jbenable=yes
;jbforce=yes
;jbmaxsize=20
;jbresyncthreshold=1000
tos=0×10
directmedia=no
nat=no
directrtpsetup=no

[provider]
type=peer
host=XXX.XXX.X.X
insecure=port,invite
context=XXXXXXXXX
dtmfmode=rfc2833
directmedia=no
nat=no
directrtpsetup=no
;qualify=300

[one-phone]
[10]
type=peer
context=XXXXXXXXX
secret=XXXXXX
host=dynamic
;qualify=300
directmedia=no
nat=no
directrtpsetup=no
dtmfmode=inband

Any help greatly appreciated!

Thanks,
Georg

Hi

I’m experimenting with using a port other than 5060 on one of our
asterisk servers.

Does anyone know how to change the target port on a Snom phone.
I have tried adding : to the end of the registrar but
this doesn’t work.
Advanced -> SIP/RTP -> Network identity(port) is something else before
anyone suggests it.

Thanks in advance

Ish

Hi All,
I am having an issue with layer 2 in BRi connection configured using Misdn:
after the BRI line working fine, a technician from our phone company came
in to add another number, after testing with his ISDN phone and BRI line is
working, from our asterisk server it is not .
When I check ports with “misdn show port x” it says:

* Port 4 Type TE Prot. PMP L2Link DOWN L1Link:UP Blocked:0 Debug:7
(as you can see here L1Link=UP *but* L2Link=DOWN)

since we didn’t change any setting from our end, it maybe an change from
our phone provider end that caused to have a *mismatch in the layer 2
connection*.

*what we tried is:*
- we did reboot our phone system several times.
- I did reconfigure the ISDN trunk setting to refresh things (*using the
same config that worked before*)
- I did use another phone system that was not used before (to eliminate the
possibility that our current phone system has hardware issues) with no
success, it *displaying the same Layer 2 issue found*.
- I turned debugging on when making calls, it shows that calls go through
as programmed but it hits a down line (because of L2 issue) :
IP0x*CLI>