[Spce-user] Spce-user Digest, Vol 8, Issue 11

Thu Jul 14 06:03:47 EDT 2011

On Thu, Jul 14, 2011 at 12:00 PM, <spce-user-request at lists.sipwise.com>wrote:

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> Today's Topics:
>
>   1. New user question... (Robert B)
>   2. Re: New user question... (Andreas Granig)
>   3. Re: Multiple extensions per user feature, in pipeline?
>      (Andreas Granig)
>   4. Re: Sipwise suddenly stopped accepting registrations.
>      (Andreas Granig)
>   5. Re: New user question... (Robert B)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 13 Jul 2011 16:05:08 -0500
> From: Robert B <devo at spudland.com>
> Subject: [Spce-user] New user question...
> To: spce-user at lists.sipwise.com
> Message-ID: <4E1E0884.5040505 at spudland.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hi folks...
>
> I'm trying to get a fairly basic setup going here to serve a fairly
> specific purpose that, on its surface, seems like a fairly trivial task.
>
> I've got a few IP-PBX systems setup for clients and friends and I
> maintain relationships with a few different wholesalers. The problem is
> management of these IP-PBXes when IP addresses change. So I'm trying to
> build my own SIP gateway where I can control the dialplans in a central
> local, then provide clients with my own server's hostname for its gateway.
>
> I am not interested in dealing with media streams. I'm leaving that up
> to the endpoint devices. I don't run Asterisk and I don't want to be in
> the B2BUA business.
>
> So I have the software going in an EC2 instance. I have my softphone
> registered successfully. I've created a SIP peering contract and added a
> SIP peer's IP and port.
>
> When I try to place a call to a PTSN number, I receive a "service or
> option unavailable" response and the Kamailio-lb.log says:
>
> Jul 13 21:01:11 spce /usr/sbin/kamailio[3178]: INFO: <script>: New
> request - M=INVITE
> R=sip:4025551212 at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> F=sip:devo at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> T=sip:4025551212 at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> IP=my.ip.ad.dr:1039 ID=XYZ.
> Jul 13 21:01:11 spce /usr/sbin/kamailio[3178]: INFO: <script>: NATed
> request detected - M=INVITE
> R=sip:4025551212 at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> F=sip:devo at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> T=sip:4025551212 at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> IP=my.ip.ad.dr:1039 ID=XYZ.
> Jul 13 21:01:11 spce /usr/sbin/kamailio[3178]: INFO: <script>: Relaying
> request, du='sip:127.0.0.1:5062' - M=INVITE
> R=sip:4025551212 at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> F=sip:devo at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> T=sip:4025551212 at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> IP=my.ip.ad.dr:1039 ID=XYZ.
> Jul 13 21:01:14 spce /usr/sbin/kamailio[3195]: WARNING: <script>:
> Inbound failover - M=INVITE
> R=sip:4025551212 at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> F=sip:devo at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> T=sip:4025551212 at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> IP=my.ip.ad.dr:1039 ID=XYZ.
> Jul 13 21:01:14 spce /usr/sbin/kamailio[3195]: ERROR: <script>: Failed
> to select next proxy - M=INVITE
> R=sip:4025551212 at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> F=sip:devo at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> T=sip:4025551212 at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> IP=my.ip.ad.dr:1039 ID=XYZ.
> Jul 13 21:01:14 spce /usr/sbin/kamailio[3180]: INFO: <script>: New
> request - M=ACK
> R=sip:4025551212 at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> F=sip:devo at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> T=sip:4025551212 at ec2-abcdefg.compute.amazonaws.com;transport=UDP
> IP=my.ip.ad.dr:1039 ID=XYZ.
>
> Any ideas what I'm not doing right here?
>
> -- Robert
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Wed, 13 Jul 2011 23:17:16 +0200
> From: Andreas Granig <agranig at sipwise.com>
> Subject: Re: [Spce-user] New user question...
> To: spce-user at lists.sipwise.com
> Message-ID: <4E1E0B5C.1070109 at sipwise.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Robert,
>
> On 07/13/2011 11:05 PM, Robert B wrote:
> > When I try to place a call to a PTSN number, I receive a "service or
> > option unavailable" response and the Kamailio-lb.log says:
>
> Looks like kamailio-proxy is not reachable. Check with
> "/etc/init.d/kamailio-proxy status" and "netstat -plen|grep 5062" if
> it's really running, and also when starting with
> "/etc/init.d/kamailio-proxy start" check the kamailio-proxy.log.
>
> Andreas
>
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> ------------------------------
>
> Message: 3
> Date: Wed, 13 Jul 2011 23:19:30 +0200
> From: Andreas Granig <agranig at sipwise.com>
> Subject: Re: [Spce-user] Multiple extensions per user feature, in
>        pipeline?
> To: spce-user at lists.sipwise.com
> Message-ID: <4E1E0BE2.7070700 at sipwise.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Henrik,
>
> On 07/13/2011 09:10 AM, Henrik Aagaard S?rensen wrote:
> > So a user have several clients (with different SIP URI) and the first
> > that accepts the call gets it.
>
> Sorry, there are no plans yet to reach with one DID different
> subscribers. What you can do is register multiple clients for one
> subscriber.
>
> Andreas
>

Hi Andreas.

When you say "What you can do is register multiple clients for one subscriber",
how does I do this?
It's for use with PTSN numbers, so I can call several "real" phones in a
group.

>
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> ------------------------------
>
> Message: 4
> Date: Wed, 13 Jul 2011 23:21:56 +0200
> From: Andreas Granig <agranig at sipwise.com>
> Subject: Re: [Spce-user] Sipwise suddenly stopped accepting
>        registrations.
> To: spce-user at lists.sipwise.com
> Message-ID: <4E1E0C74.3070109 at sipwise.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Henrik,
>
> On 07/13/2011 09:08 AM, Henrik Aagaard S?rensen wrote:
> > Jul 13 06:43:25 spce /usr/sbin/kamailio[5490]: ALERT: <core>
> > [main.c:741]: child process 5491 exited by a signal 9
>
> Looks like you ran out of memory, double-check by executing "dmesg" if
> the oom-killer hit in. If so, decrease your buffer pool size value in
> config.yml and limit the number of forked apache processes to prevent
> this from happening (or alternatively use a larger instance).
>
> Andreas
>
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> ------------------------------
>
> Message: 5
> Date: Wed, 13 Jul 2011 16:42:07 -0500
> From: Robert B <devo at spudland.com>
> Subject: Re: [Spce-user] New user question...
> To: Andreas Granig <agranig at sipwise.com>
> Cc: spce-user at lists.sipwise.com
> Message-ID: <4E1E112F.60701 at spudland.com>
> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"
>
> Andreas,
>
> That's really weird. After restarting the instance kamailio seems to be
> running. Not sure what I did that killed it the first time.
>
> So anyway, now my calls immediately go to "normal temporary failure"...
> Looking at the logs, I see the following in kamailio-proxy.log:
>
> Jul 13 21:39:08 spce /usr/sbin/kamailio[3623]: ERROR: <script>: No PSTN
> gateways available ...
>
> I do, indeed, have a valid SIP peer configured for call termination.
>
> Your help is genuinely appreciated!
>
> -- Robert
>
>
> On 7/13/2011 4:17 PM, Andreas Granig wrote:
> > Hi Robert,
> >
> > On 07/13/2011 11:05 PM, Robert B wrote:
> >> When I try to place a call to a PTSN number, I receive a "service or
> >> option unavailable" response and the Kamailio-lb.log says:
> > Looks like kamailio-proxy is not reachable. Check with
> > "/etc/init.d/kamailio-proxy status" and "netstat -plen|grep 5062" if
> > it's really running, and also when starting with
> > "/etc/init.d/kamailio-proxy start" check the kamailio-proxy.log.
> >
> > Andreas
> >
> >
> >
> > _______________________________________________
> > Spce-user mailing list
> > Spce-user at lists.sipwise.com
> > http://lists.sipwise.com/listinfo/spce-user
>
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>
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> End of Spce-user Digest, Vol 8, Issue 11
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