Situation:
Endpoint is called Party (Sipura SPA2000)
Calling Party is routed through from a TDM call and VoIP'd by Cisco 3640
router then sent to SER.
Endpoint receives call - everything works. When the Sipura hangs up the
call, it sends a BYE to ser - but ser does not send the BYE to the router to
disconnect the TDM channel.
All other situations result in a completed call with channels released on
the router just fine.
After further inspection - we noticed that when the Sipura hangs up and send
the BYE to SER, SER responds with this:
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 11.11.111.155:5060;branch=z9hG4bK-b76fe29f
From: <sip:8646783182@22.22.222.154>;tag=bc5a80577e3d5642
To: <sip:8644679887@33.33.333.157>;tag=322BCA98-25A3
Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)33.33.333.157
CSeq: 101 BYE
Contact: <sip:8644679887@44.44.444.148>
Server: Sip EXpress router (0.8.12 (i386/linux))
Content-Length: 0
Warning: 392 44.44.444.148:5060 "Noisy feedback tells: pid=20033
req_src_ip=22.22.222.154 req_src_port=5060
in_uri=sip:8644679887@44.44.444.148:5060
out_uri=sip:8644679887@44.44.444.148:5060 via_cnt==0"
Any ideas?
Thanks,
- Darren
Hello All,
First, I created subtle problems in SER by having an alias entry for the host ip in the ser.cfg file.
I don't recommend doing the following [where 10.10.10.49 is the ip of the host where SER is running].
alias=10.10.10.49
Next, I've had similar problems to a user who posted recently about SEMS being unstable. Occassionally it just stops playing the greeting, however, even if the greeting doesn't play, SEMS will still record audio. Overall, I've had a lot of frustration with SEMS.
Attached is my ser.cfg file and some error messages from syslog.
Here is a scenario which is causing problems with voicemail.
------------------------
User sip:info@mydomain.com is not logged in anywhere, however that user has calls being forwarded to sip:mvestal@mydomain.com and sip:esavelle@mydomain.com.
The location table shows these entries correctly. In addition, mvestal and esavelle are also in the location table at specific ip addresses.
When a call come in for sip:info@mydomain.com, it gets forked propery. However, if no one answers the call before the fr_inv_timer expires, SER tries to hand the call off to VMSER [second instance of SER which is running on port 5090 and is handling voicemail] but the error messages in the attached logs show up. The calling party typically hears ringing continue after SER had timed out, and then gets a busy tone a few seconds later.
I'm also going to try Asterisk. There is a specific feature of Asterisk which I think would benefit SER/SEMS a lot.
We will set up Asterisk so that we do and append_branch(); in SER as soon as a call is initiated to an internal extension. Asterisk can then be set to wait a certain amount of time before it answers the call. This way, the fr_inv_time feature of SER isn't necessary for getting calls into voicemail [SEMS or Asterisk.]
Also, it is very nice to be able to use a phone to check voicemail. No all users are necessarily sitting on their email inboxes all day long.
Any help on the bug would be greatly appreciated.
Noah
---------------------------------
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Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709675@10.10.10.5>;tag=4979781C-1816, i-uri=
sip:esavelle@mydomain.com, method=INVITE, o-uri=sip:esavelle@10.10.10.167:5060, code=408 Request Timeout
Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: failure_route[4]
Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail
Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709675@10.10.10.5>;tag=4979781C-1816, i-uri=
sip:mvestal@mydomain.com, method=INVITE, o-uri=sip:mvestal@10.10.10.98:5060, code=408 Request Timeout
Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: failure_route[4]
Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24242]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487
Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24229]: ACC: request acknowledged: from=<sip:4083709675@10.10.10.5>;tag=4979781C-181
6, i-uri=sip:esavelle@mydomain.com, method=ACK, o-uri=sip:esavelle@10.10.10.167:5060, code=180
Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail
Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709675@10.10.10.5>;tag=4979781C-1816, i-uri=
sip:6600@10.10.10.49:5060, method=INVITE, o-uri=sip:mvestal@mydomain.com, code=408 Request Timeout
Dec 16 16:14:02 jiffypop /usr/local/sbin/vmser[29524]: t_newtran created
Dec 16 16:14:02 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - begin ******************
Dec 16 16:14:02 jiffypop /usr/local/sbin/vmser[29523]: t_newtran created
Dec 16 16:14:02 jiffypop /usr/local/sbin/vmser[29523]: **************** vm start - begin ******************
Dec 16 16:14:02 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - end ******************
Dec 16 16:14:02 jiffypop /usr/local/sbin/vmser[29523]: **************** vm start - end ******************
Dec 16 16:14:02 jiffypop Sems[19578]: Error: 482 Loop detected
Dec 16 16:14:41 jiffypop /usr/local/sbin/ser[24240]: ACC: transaction answered: from="bkern" <sip:bkern@mydomain.com>;tag=9f5e9
214-b2f7-404c-a109-e20502c672c7, i-uri=sip:fchuang@mydomain.com, method=SUBSCRIBE, o-uri=sip:10.10.10.192:13584, code=200
Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709725@10.10.10.5>;tag=497A2F34-544, i-uri=s
ip:mvestal@mydomain.com, method=INVITE, o-uri=sip:mvestal@10.10.10.98:5060, code=408 Request Timeout
Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: failure_route[4]
Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail
Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709725@10.10.10.5>;tag=497A2F34-544, i-uri=s
ip:esavelle@mydomain.com, method=INVITE, o-uri=sip:esavelle@10.10.10.167:5060, code=408 Request Timeout
Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: failure_route[4]
Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24231]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487
Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24229]: ACC: request acknowledged: from=<sip:4083709725@10.10.10.5>;tag=497A2F34-544
, i-uri=sip:mvestal@mydomain.com, method=ACK, o-uri=sip:mvestal@10.10.10.98:5060, code=180
Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail
Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709725@10.10.10.5>;tag=497A2F34-544, i-uri=s
ip:6600@10.10.10.49:5060, method=INVITE, o-uri=sip:mvestal@mydomain.com, code=408 Request Timeout
Dec 16 16:14:49 jiffypop /usr/local/sbin/vmser[29524]: t_newtran created
Dec 16 16:14:49 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - begin ******************
Dec 16 16:14:49 jiffypop /usr/local/sbin/vmser[29523]: t_newtran created
Dec 16 16:14:49 jiffypop /usr/local/sbin/vmser[29523]: **************** vm start - begin ******************
Dec 16 16:14:49 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - end ******************
Dec 16 16:14:49 jiffypop /usr/local/sbin/vmser[29523]: **************** vm start - end ******************
Dec 16 16:14:49 jiffypop Sems[19578]: Error: 482 Loop detected
Dec 16 16:15:10 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:5107392224@10.10.10.5>;tag=497A8154-2635, i-uri=
sip:6611@10.10.10.49:5060, method=INVITE, o-uri=sip:service@10.10.10.242:5060, code=408 Request Timeout
Dec 16 16:15:10 jiffypop /usr/local/sbin/ser[24298]: failure_route[4]
Dec 16 16:15:10 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail
Dec 16 16:15:10 jiffypop /usr/local/sbin/ser[24233]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487
Dec 16 16:15:10 jiffypop /usr/local/sbin/vmser[29524]: t_newtran created
Dec 16 16:15:10 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - begin ******************
Dec 16 16:15:10 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - end ******************
Dec 16 16:15:18 jiffypop /usr/local/sbin/vmser[29523]: t_newtran created
Dec 16 16:15:18 jiffypop /usr/local/sbin/vmser[29523]: **************** vm end - begin ******************
Dec 16 16:15:18 jiffypop /usr/local/sbin/vmser[29523]: **************** vm end - end ******************
Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709675@10.10.10.5>;tag=497AC670-18B, i-uri=s
ip:mvestal@mydomain.com, method=INVITE, o-uri=sip:mvestal@10.10.10.98:5060, code=408 Request Timeout
Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: failure_route[4]
Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail
Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709675@10.10.10.5>;tag=497AC670-18B, i-uri=s
ip:esavelle@mydomain.com, method=INVITE, o-uri=sip:esavelle@10.10.10.167:5060, code=408 Request Timeout
Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: failure_route[4]
Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24240]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487
Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24229]: ACC: request acknowledged: from=<sip:4083709675@10.10.10.5>;tag=497AC670-18B
, i-uri=sip:mvestal@mydomain.com, method=ACK, o-uri=sip:mvestal@10.10.10.98:5060, code=180
Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail
Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709675@10.10.10.5>;tag=497AC670-18B, i-uri=s
ip:6600@10.10.10.49:5060, method=INVITE, o-uri=sip:mvestal@mydomain.com, code=408 Request Timeout
Dec 16 16:15:28 jiffypop /usr/local/sbin/vmser[29524]: t_newtran created
Dec 16 16:15:28 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - begin ******************
Dec 16 16:15:28 jiffypop /usr/local/sbin/vmser[29525]: t_newtran created
Dec 16 16:15:28 jiffypop /usr/local/sbin/vmser[29525]: **************** vm start - begin ******************
Dec 16 16:15:28 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - end ******************
Dec 16 16:15:28 jiffypop /usr/local/sbin/vmser[29525]: **************** vm start - end ******************
Dec 16 16:15:28 jiffypop Sems[19578]: Error: 482 Loop detected
Dec 16 16:15:31 jiffypop Sems[19578]: Error: while getting return code from Ser
Hi,
Can I use asterisk as a voicemail server, where unanswered calls get
forwarded to asterisk acting as a voicemail server.
I like to fact that asterisk voicemail can be accessed through a phone
(IVR).
Is this setup feasable, has anyone tried it ?
Thanks.
Samy.
Sipura Firmware is 1.0.20
We are already in talks with Sipura about this issue as we found a bug for
them yesterday. It appears though based on the Etheral traces and ngrep that
the Sipura is sending the BYE messages - but SER is not forwarding them
along.
We are using Ser 8.12 on a Sun Ultra 60 - and as I said before, all other
instances work flawlessly.
Thanks,
- Darren
-----Original Message-----
From: Andres [mailto:andres@telesip.net]
Sent: Thursday, December 18, 2003 2:58 PM
To: Darren Sessions; serusers(a)lists.iptel.org
Subject: Re: [Serusers] BYE Message Problem
On Thursday 18 December 2003 14:50, Darren Sessions wrote:
> We also have a problem with BYE's not being sent from an on-net call
> (sipura to sipura).
What firmware version are you using? We have been testing the SPA2000 for
about 3 weeks now and have not seen this issue.
>
> -----Original Message-----
> From: Darren Sessions
> Sent: Thursday, December 18, 2003 2:32 PM
> To: 'serusers(a)lists.iptel.org'
> Subject: RE: Serusers Digest, Vol 8, Issue 16
> Importance: High
>
>
> Situation:
>
> Endpoint is called Party (Sipura SPA2000)
>
> Calling Party is routed through from a TDM call and VoIP'd by Cisco 3640
> router then sent to SER.
>
> Endpoint receives call - everything works. When the Sipura hangs up the
> call, it sends a BYE to ser - but ser does not send the BYE to the router
> to disconnect the TDM channel.
>
> All other situations result in a completed call with channels released on
> the router just fine.
>
> After further inspection - we noticed that when the Sipura hangs up and
> send the BYE to SER, SER responds with this:
>
>
> SIP/2.0 481 Call/Transaction Does Not Exist
> Via: SIP/2.0/UDP 11.11.111.155:5060;branch=z9hG4bK-b76fe29f
> From: <sip:8646783182@22.22.222.154>;tag=bc5a80577e3d5642
> To: <sip:8644679887@33.33.333.157>;tag=322BCA98-25A3
> Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)33.33.333.157
> CSeq: 101 BYE
> Contact: <sip:8644679887@44.44.444.148>
> Server: Sip EXpress router (0.8.12 (i386/linux))
> Content-Length: 0
> Warning: 392 44.44.444.148:5060 "Noisy feedback tells: pid=20033
> req_src_ip=22.22.222.154 req_src_port=5060
> in_uri=sip:8644679887@44.44.444.148:5060
> out_uri=sip:8644679887@44.44.444.148:5060 via_cnt==0"
>
>
> Any ideas?
>
> Thanks,
>
> - Darren
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
We also have a problem with BYE's not being sent from an on-net call (sipura
to sipura).
-----Original Message-----
From: Darren Sessions
Sent: Thursday, December 18, 2003 2:32 PM
To: 'serusers(a)lists.iptel.org'
Subject: RE: Serusers Digest, Vol 8, Issue 16
Importance: High
Situation:
Endpoint is called Party (Sipura SPA2000)
Calling Party is routed through from a TDM call and VoIP'd by Cisco 3640
router then sent to SER.
Endpoint receives call - everything works. When the Sipura hangs up the
call, it sends a BYE to ser - but ser does not send the BYE to the router to
disconnect the TDM channel.
All other situations result in a completed call with channels released on
the router just fine.
After further inspection - we noticed that when the Sipura hangs up and send
the BYE to SER, SER responds with this:
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 11.11.111.155:5060;branch=z9hG4bK-b76fe29f
From: <sip:8646783182@22.22.222.154>;tag=bc5a80577e3d5642
To: <sip:8644679887@33.33.333.157>;tag=322BCA98-25A3
Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)33.33.333.157
CSeq: 101 BYE
Contact: <sip:8644679887@44.44.444.148>
Server: Sip EXpress router (0.8.12 (i386/linux))
Content-Length: 0
Warning: 392 44.44.444.148:5060 "Noisy feedback tells: pid=20033
req_src_ip=22.22.222.154 req_src_port=5060
in_uri=sip:8644679887@44.44.444.148:5060
out_uri=sip:8644679887@44.44.444.148:5060 via_cnt==0"
Any ideas?
Thanks,
- Darren
I can reproduce the choppy audio
setup:
budgetone-100 <----> ser+rtpproxy(version from today) on redhat 9 (or 8,
not sure) <------>x-lite build 1088 on win xp
the choppy sound occours every 5 seconds, in both directions.
RTP-analysis:
RTP stream from budgetone to x-lite (via rtpproxy) is fine (no jitter,
no loss)
RTP stream from x-lite to budgetone (via rtpproxy): very strange: x-lite
switches to a new SSRC (and seq-nr. start at 0 again) every 5 seconds.
so, in my opinion: rtpproxy works fine, x-lite is guilty. but why does
this happen only if the rtp proxy is involved? ...further investigations
are planned...
Klaus
> -----Original Message-----
> From: Jan Janak [mailto:janak@fokus.fraunhofer.de]
> Sent: Wednesday, December 10, 2003 4:09 PM
> To: Adrian Georgescu
> Cc: serusers(a)lists.iptel.org; Ricardo Villa
> Subject: Re: [Serusers] Re: Xten-RTPProxy choppy audio
>
>
> Well, we can generate the traces locally, but I haven't encounter the
> problem you describe so it makes no sense.
>
> You wrote you are able to reproduce the problem, in that case I would
> like to ask you to generate the traces that show the problem
> so we could
> analyze and fix it.
>
> Also please tell us on what OS does this happen ? (I mean the
> OS the proxy
> is running on).
>
> Another question, what is the license of your RTP proxy ? I
> didn't find
> any licensing info in the sources. Will the sources be available ?
>
> Jan.
>
> On 10-12 16:04, Adrian Georgescu wrote:
> > On Wednesday, Dec 10, 2003, at 15:48 Europe/Amsterdam,
> Ricardo Villa
> > wrote:
> >
> > >Adrian,
> > >
> > >Do you have an Etheral trace trace of such a call (using G711)?
> >
> > I guess ethereal traces can be generated with ethereal program and
> > decoded locally on your servers if you want to isolate this.
> >
> > > I can
> > >decode it and produce an audio file for all to examine.
> This way we
> > >can get
> > >to the bottom of this.
> > >Thanks,
> > >Ricardo
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
This is only the forwarding code.
You need to merge with the default configuration code which came with
asterisk.
Kannaiyan
----- Original Message -----
From: "Samy Touati" <samy(a)tunix.com>
To: "Kannaiyan Natesan" <nkans(a)lycos.co.uk>
Sent: Thursday, December 18, 2003 5:15 PM
Subject: Re: [Serusers] Using asterisk as a voicemail server with SER
> Hi,
>
> Sorry to bother on this, but I'm trying to understand how all of his
> work.
> In the code snippet, you sent me ser is redirecting the call to
> asterisk. Are the phones registering in SIP to asterisk or to ser ?
> I want the phones to register with ser but have mailboxes on asterisk.
> Is this supported in asterisk, are you doing this?
>
> Thanks again.
>
> Samy.
>
>
> ----- Original Message -----
> From: "Kannaiyan Natesan" <nkans(a)lycos.co.uk>
> Date: Monday, December 15, 2003 2:54 pm
> Subject: Re: [Serusers] Using asterisk as a voicemail server with SER
>
> > Samy,
> >
> > Hope you find this useful.
> > here is the snippet
> >
> > # ------------------ module loading -------------------------------
> > ---
> >
> > ... your other modules here
> >
> > loadmodule "modules/sl/sl.so"
> > loadmodule "modules/tm/tm.so"
> >
> > # ----------------- module parameters ---------------
> >
> > # -- tm params --
> > # fr_inv_timer sets value for INVITE transactions
> > # fr_timer for all others
> >
> > modparam("tm", "fr_inv_timer", 15 )
> > modparam("tm", "fr_timer", 10 )
> >
> > #-------------------------------------------------------
> >
> > route{
> >
> > .........
> >
> > append_branch("sip:asterisk@asthost:portno"); # tell your
> > asterisk URI here
> > # forward if the call fails
> > t_on_failure("1");
> > # start forwarding all calls now
> > t_relay();
> >
> > }
> >
> > failure_route[1] {
> > log(1,"asterisk voicemail\n");
> > t_relay();
> > }
> >
> > Kannaiyan
> >
> > ----- Original Message -----
> > From: "Samy Touati" <samy(a)tunix.com>
> > To: "Kannaiyan Natesan" <nkans(a)lycos.co.uk>
> > Cc: <serusers(a)lists.iptel.org>
> > Sent: Monday, December 15, 2003 7:45 PM
> > Subject: Re: [Serusers] Using asterisk as a voicemail server with SER
> >
> >
> > > Only one user on asterisk will collect all voice mails.
> > > I was thinking that I will need to create all the extensions in
> > > asterisk so when the user cannot be reached at his phone, then ser
> > > will rewrite the request to asterisk keeping the original uri.
> > > Is this how to do it ?
> > >
> > > Thanks.
> > > Samy.
> > >
> > >
> > >
> > > ----- Original Message -----
> > > From: "Kannaiyan Natesan" <nkans(a)lycos.co.uk>
> > > Date: Monday, December 15, 2003 2:41 pm
> > > Subject: Re: [Serusers] Using asterisk as a voicemail server
> > with SER
> > >
> > > > Samy,
> > > >
> > > > That is feasible.
> > > > You can create a sip user in asterisk and forward the call
> > on busy.
> > > >
> > > > Kannaiyan
> > > >
> > > >
> > > > ----- Original Message -----
> > > > From: "Samy Touati" <samy(a)tunix.com>
> > > > To: <serusers(a)lists.iptel.org>
> > > > Sent: Monday, December 15, 2003 7:35 PM
> > > > Subject: [Serusers] Using asterisk as a voicemail server with SER
> > > >
> > > >
> > > > > Hi,
> > > > >
> > > > > Can I use asterisk as a voicemail server, where unanswered
> calls
> > > > get
> > > > > forwarded to asterisk acting as a voicemail server.
> > > > > I like to fact that asterisk voicemail can be accessed
> > through a
> > > > phone
> > > > > (IVR).
> > > > > Is this setup feasable, has anyone tried it ?
> > > > >
> > > > > Thanks.
> > > > >
> > > > > Samy.
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Serusers mailing list
> > > > > serusers(a)lists.iptel.org
> > > > > http://lists.iptel.org/mailman/listinfo/serusers
> > > > >
> > > > >
> > > >
> > > >
> > > >
> > > >
> > >
> > >
> >
> >
> >
> >
> >
>
>
