Hi all,
I'm trying to find my way around the ser configuration and the sip protocol
in general, so please forgive me if this is a faq (I can't find answers in
the list archives).
I need to enforce the user agents to pick only their registered
aliases/usernames instead of being able to pick an arbitrary contact.
Ideally, I would like to have an aphanumeric account for each user agent and
a fixed numeric alias stored in a database. If the user agent tried to
register with another alias/contact instead of the one stored in the
database, it should be denied.
How do I implement something similar ?
Thank you,
Dave.
Hi all. I was hoping someone could double-check my thinking, and make
sure I get off on the right foot with a new SER setup. I don't think
any of the examples in the admin manual cover my scenario, so I'm not
entirely sure on how to approach it.
Here comes the crude ASCII:
+--------+ +-------------------+ +-------+ +----------------------+
| Phones |___| Router/SER Server |___| I'net |___| SER Server/PSTN Gate |
| 10.x | | 10.0.0.1 - Public | | * | | 17.40.2.42 |
+--------+ +-------------------+ +-------+ +----------------------+
[Figure explanation: The phones (Cisco 7960s) are on a private, NAT'd
network. The IP router for this network also happens to be the local
SIP server, running SER. Another SER server, reachable via the
Internet, has access to a PSTN Gateway.]
All of the documentation I've seen assumes that either the phones are
behind NAT, and the SIP server is outside, or that both devices are
public. I'm hoping that having the SER server multihomed will ease
some of the issues associated with SIP-through-NAT. Testing without
the local server showed problems with call transferring and the like,
though I still suspect this was entirely my fault.
What would be the proper way to go about configuring SER for this
type of network layout? If I use record_route() and proxy all of the
RTP traffic, does this avoid needing to mangling up the poor packets
with the nathelper module? It seems like if I add "mhomed=1" to the
local server, to get the "Via" header set to the external interface's
address, and add "reply_to_via=yes" on the public SIP server, the
local SER should be able to function like a traditional proxy for the
phones. Is this the next-best thing to end-to-end connectivity?
How about user configuration? Add accounts for each of phone numbers
to both SER servers, grant them to a new group on the public server
(17.40.2.42), and then, to forward inbound calls, is it as simple as:
is_user_in("To", "my-local-group") { route(x) }
If it's any easier, I *can* move the SER server to a separate machine
behind the NAT on the 10.x network.
Thanks for listening, and extra thanks for any insight you can provide.
Jeremy
--
Jeremy M. Dolan <mailto:jmd@pobox.com> <http://jmd.us/>
PGP: 1024D/3C68A1BA 9470 210C A476 FFBB 6D11 0223 0D1C ABFC 3C68 A1BA
HI,
WE are having problems with t.38 fax and record-route in SER.
Without record_route, using only t_relay(), fax works fine between two
of our UA's.
However, when record_route is used, a voice call is successful but a fax
call fails. An ethereal sniff shows that the "re-invite" from the
destination to the origination is never forwarded to the origination
UA. Thus, the fax fails.
If I turn off record_route(), the fax call succeeds.
An ethereal sniff of the failure case is attached. A snip of the ser.cfg
file which shows the use of record_route() and t_relay() is attached.
If I comment out the one line for record_route(), the fax succeeds.
Any ideas as to why this fails?
thanks
bert
--
Bert Berlin
Director, System Test
Quintum Technologies, Inc.
71 James Way
Eatontown,NJ 07724
ph 732-460-9000 ext 247
if (uri==myself)
{
#look for the registered contact in the location table of database
#and if find it, rewrite the uri and forward statefully to the destination
if (lookup ("location"))
{
record_route();#send everything back through the proxy
t_relay();
break;
}
Hi
I am trying to send all my calls to a PSTN gateway (asterisk with digium
cards running on the same machine on 7060). I can use forward, but then I
cannot use the solution SER provides me by using NAT helper. Consider the
forwarding script:
      if (uri=~"^sip:[8].*@")
{
forward(209.7.34.58,7060);
    }
Here it works fine if I am calling from a public IP, but will not work if
I am calling from a NATed client since this stateless forwarding will
take SER out of the picture, and rtpproxy will not be used. I have also
tried using t_relay_to_udp, but is is doing the same, the below lines are
still not making SER to proxy the request.
if (uri=~"^sip:[8].*@")
{
t_relay_to_udp("209.7.34.58","7060");
break;
}
What would be the best way of doing it. I want SER to act as a proxy
(B2BUA?) between my caller and the PSTN asterisk server. I want to be able
to use nathelper for this scenario.
Regards
HI,
I am using Version 12.3(7)T1. I am doing very simple thing, routing the
call from a sip phone via ser to PSTN (cisco 5400). Right now, I have
send call via h323 from excel switch to cisco without any problem. But
when ever I am sending call to excel switch, it hangs.
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
C.M. Rahman Jr.
CTO
CCNP, MCSE Security "Secure your self by securing your System"
CompTI Security Plus Certified
CCS Internet
http://www.ccsi.com
13704 Research Blvd. Building O-Suite 4
Austin, TX 78750
Tel: 512-257-2274 Ex: 115
-----Original Message-----
From: Steve Blair [mailto:blairs@isc.upenn.edu]
Sent: Monday, June 28, 2004 5:13 PM
To: CM Rahman
Cc: Stephen Kingham; serusers(a)lists.iptel.org
Subject: Re: [Serusers] as5400 and ser
What version of IOS are you using? I have a similar issue with a
Verizon DMS100 switch right now. It turns out that some 12.2
versions of IOS have an "incompatibility" with certain switches.
Cisco does not consider this a bug. They say that it is a difference
in the interpretation of the signaling standard.
In either case upgrading to 12.3 releases is suppose to fix the
problem according to Cisco. Instead I found that it improves but
doesn't necessarily fix the problem. Here is my story.
I am trying to use the CC-Diversion header so that when an
inbound call (to an IP phone) is not answered the call is redirected
out through the gateway to the DMS100 which then has an SMDI
link into our Octel 350 VM ystem.
The SER part has been working. Initially the redirected call
just hung, dead air, until I hung up the phone. You could see this
in the debug messages on the Cisco.
When I upgraded to 12.3.9 main line release I got the general
voice mail greeting regardless of which phone initiated the call.
Being the difficult person that I am I downgraded to a 12.3 T
train
release to see what happened. Now if I call from my Centrex phone
on my desk I get the greeting associated with the Calling Party ID,
my Centrex phone. If I call from a non-Penn number however I get
the general voice mail greeting. Cisco has yet to explain what is
happening
but they continue to claim the problem is resolved.
The one difference is that none of the releases I upgraded to are
listed on the feature report that Cisco published, however, I have not
yet
been able to get one of the identified releases. The case is still open.
Then again as soon as we brought this to Verizon's attention they
tested the PRIs and said Oh, you aren't paying for voice mail service
on that trunk. We are waiting for them to determine if we have to
pay an additional fee for this feature.
I don't have the Cisco case in front of me but what you should look
into is how the called number is mapped to the Redirect Information
Element field. If you get stuck drop me another note and I'll see if
I can dig up the specific cae.
Good luck,
Steve
CM Rahman wrote:
>Actually I have a Lucent Excel switch which is connected to the cisco
>as5400 via T1 Pri. Anybody here using Excel switch with a cisco ?
>
>Right now, when ever I do debug q931 I get this below and it hangs
until
>my messenger times out and it disconnects. It should answer and give me
>voice prompt. Anybody have deal with same scenario as mine?
>
>
>
>*Feb 18 15:40:20.142: ISDN Se7/0:3:23 Q931: Applying typeplan for
>sw-type 0xD is 0x2 0x1, Called num 5122200090
>*Feb 18 15:40:20.142: ISDN Se7/0:3:23 Q931: TX -> SETUP pd = 8 callref
>= 0x005F
> Bearer Capability i = 0x8090A2
> Standard = CCITT
> Transer Capability = Speech
> Transfer Mode = Circuit
> Transfer Rate = 64 kbit/s
> Channel ID i = 0xA98381
> Exclusive, Channel 1
> Called Party Number i = 0xA1, '5122200090'
> Plan:ISDN, Type:National
>*Feb 18 15:40:20.158: ISDN Se7/0:3:23 Q931: RX <- CALL_PROC pd = 8
>callref = 0x805F
> Channel ID i = 0xA98381
> Exclusive, Channel 1
>
>&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
>C.M. Rahman Jr.
>CTO
>CCNP, MCSE Security "Secure your self by securing your System"
>CompTI Security Plus Certified
>CCS Internet
>http://www.ccsi.com
>13704 Research Blvd. Building O-Suite 4
>Austin, TX 78750
>Tel: 512-257-2274 Ex: 115
>
>-----Original Message-----
>From: Stephen Kingham [mailto:Stephen.Kingham@aarnet.edu.au]
>Sent: Monday, June 28, 2004 5:50 AM
>To: CM Rahman
>Cc: Richard; serusers(a)lists.iptel.org
>Subject: Re: [Serusers] as5400 and ser
>
>
>
>CM Rahman wrote:
>
>
>
>>I am sorry, I didn't show how put the pot in my last email, here it
is,
>>
>>dial-peer voice 150 voip
>>description CCSi voip phone
>>destination-pattern 9T
>>progress_ind setup enable 3
>>session protocol sipv2
>>session target ipv4:216.236.160.11
>>codec g723r53
>>
>>
>>
>>Answer to your question, without putting "isdn protocol-emulate
>>
>>
>network"
>
>
>>I wasn't able to get PRI Layer 2 up.
>>
>>
>>
>>
>Yes. ISDN has a network side and a user side so that the layer 2
>protocol Q921/lapb will work.
>
>Most PABX want to be the user side.
>
>
>
>>Any other suggestion?
>>
>>
>>
>>
>
>yes you have to have a pots dialpeer, the Cisco VoIP gateway requires
at
>
>least one, I think maybe one for each E1 port.
>
>Take a look at the template I have posted here:
>http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworks
h
>op/uas/ciscoVoIPGateways/as5300-12.3-6b-sip.txt
>
>
>
>>&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
>>C.M. Rahman Jr.
>>CTO
>>CCNP, MCSE Security "Secure your self by securing your System"
>>CompTI Security Plus Certified
>>CCS Internet
>>http://www.ccsi.com
>>13704 Research Blvd. Building O-Suite 4
>>Austin, TX 78750
>>Tel: 512-257-2274 Ex: 115
>>
>>-----Original Message-----
>>From: Richard [mailto:mypop3mail@yahoo.com]
>>Sent: Friday, June 25, 2004 4:11 PM
>>To: CM Rahman; serusers(a)lists.iptel.org
>>Subject: RE: [Serusers] as5400 and ser
>>
>>Don't know why you have the following two lines,
>>isdn protocol-emulate network
>>isdn incoming-voice modem
>>
>>Also you probably need a pots dial-peer...
>>
>>Cisco web site has some configuration samples.
>>
>>--- CM Rahman <cmrahman(a)ccsi.com> wrote:
>>
>>
>>
>>
>>>Once I send a call via messenger, I don't hear
>>>anything other side. But
>>>after a while it disconnect.
>>>
>>>Here are the cisco config
>>>
>>>******************************
>>>controller T1 7/0:3
>>>framing esf
>>>pri-group timeslots 1-24
>>>description Prism Test
>>>
>>>***************************************
>>>interface Serial7/0:3:23
>>>no ip address
>>>isdn switch-type primary-ni
>>>isdn protocol-emulate network
>>>isdn incoming-voice modem
>>>isdn T310 180000
>>>no cdp enable
>>>!***************************************
>>>
>>>dial-peer voice 150 voip
>>>description CCSi voip phone
>>>destination-pattern 9T
>>>session protocol sipv2
>>>session target ipv4:216.236.160.11
>>>codec g723r53
>>>
>>>*****************************************
>>>
>>>
>>>
>>>
>>>*Feb 15 16:18:09.720: ISDN Se7/0:3:23 Q931: Applying
>>>typeplan for
>>>sw-type 0xD is 0x2 0x1, Called num 5122200090
>>>*Feb 15 16:18:09.720: ISDN Se7/0:3:23 Q931: TX ->
>>>SETUP pd = 8 callref
>>>= 0x002E
>>> Bearer Capability i = 0x8090A2
>>> Standard = CCITT
>>> Transer Capability = Speech
>>> Transfer Mode = Circuit
>>> Transfer Rate = 64 kbit/s
>>> Channel ID i = 0xA98381
>>> Exclusive, Channel 1
>>> Called Party Number i = 0xA1, '5122200090'
>>> Plan:ISDN, Type:National
>>>*Feb 15 16:18:09.732: ISDN Se7/0:3:23 Q931: RX <-
>>>CALL_PROC pd = 8
>>>callref = 0x802E
>>> Channel ID i = 0xA98381
>>> Exclusive, Channel 1
>>>*Feb 15 16:20:17.967: ISDN Se7/0:3:23 Q931: TX ->
>>>DISCONNECT pd = 8
>>>callref = 0x002E
>>> Cause i = 0x8290 - Normal call clearing
>>>*Feb 15 16:20:17.995: ISDN Se7/0:3:23 Q931: RX <-
>>>RELEASE pd = 8
>>>callref = 0x802E
>>>*Feb 15 16:20:17.995: ISDN Se7/0:3:23 Q931: TX ->
>>>RELEASE_COMP pd = 8
>>>callref = 0x002E
>>>
>>>
>>>
>>>
>>>
>>>
>>&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
>>
>>
>>
>>
>>>C.M. Rahman Jr.
>>>CCNP, MCSE Security "Secure your self by securing
>>>your System"
>>>CompTI Security Plus Certified
>>>CCS Internet
>>>http://www.ccsi.com
>>>13704 Research Blvd. Building O-Suite 4
>>>Austin, TX 78750
>>>Tel: 512-257-2274 Ex: 115
>>>
>>>-----Original Message-----
>>>From: serusers-bounces(a)lists.iptel.org
>>>[mailto:serusers-bounces@lists.iptel.org] On
>>>Behalf Of Richard
>>>Sent: Friday, June 25, 2004 3:27 AM
>>>To: serusers(a)lists.iptel.org
>>>Subject: RE: [Serusers] as5400 and ser
>>>
>>>If you check this page,
>>>
>>>
>>>
>>>
>>>
>>>
>>http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration
_
>>
>>
>g
>
>
>>
>>
>>
>>
>>>uide_chapter09186a00800eadfa.html
>>>
>>>PSTN error "63 Service or option unavailable" is
>>>mapped to sip error "503 Service or option
>>>unavailable" which is in the header of the message.
>>>
>>>Also the page shows why IP phone or PSTN generates
>>>this and how proxy is supposed to do with it. Quote,
>>>"The SIP gateway generates this response if it is
>>>unable to process the request due to an overload or
>>>maintenance problem. Upon receiving this response,
>>>the
>>>gateway initiates a graceful call disconnect and
>>>clears the call. "
>>>
>>>Look like a pstn config issue. Use "debug isdn
>>>q931",
>>>"debug isdn q921" and "term mon" for further
>>>debuging.
>>>
>>>Cheers,
>>>Richard
>>>
>>>--- CM Rahman <cmrahman(a)ccsi.com> wrote:
>>>
>>>
>>>
>>>
>>>>Looking through your cisco config file, I am
>>>>guessing your E1 are not
>>>>Pri. Ami I correct? I am dealing with a
>>>>
>>>>
>>>>
>>>>
>>>channelized
>>>
>>>
>>>
>>>
>>>>DS3 with T1 Pri. I
>>>>will also share my config file after I can get the
>>>>call routed.
>>>>Currently I am getting this below. My
>>>>
>>>>
>>>>
>>>>
>>>understanding
>>>
>>>
>>>
>>>
>>>>is there is
>>>>something wrong in the call going from cisco to
>>>>
>>>>
>>>>
>>>>
>>>Pri
>>>
>>>
>>>
>>>
>>>>trunk. Anybody can
>>>>give me some clue, that will be great.
>>>>
>>>>
>>>>
>>>>146.82.136.218:5060 -> 216.236.160.11:5060
>>>> SIP/2.0 503 Service Unavailable..Via:
>>>>
>>>>
>>>>
>>>>
>>>SIP/2.0/UDP
>>>
>>>
>>>
>>>
>>>>216.236.160.11;branch=z9h
>>>> G4bKc513.1c338976.0,SIP/2.0/UDP
>>>>65.70.207.66:8675..From:
>>>>"pappusip(a)backup.c
>>>> csi.com"
>>>>
>>>>
>>>>
>>>>
>>>>
>><sip:pappusip@backup.ccsi.com>;tag=c270cb2a9ab14343b72218adb808612
>>
>>
>>
>>
>>>> 4;epid=c91b05026b..To:
>>>>
>>>>
>>>>
>>>>
>>>>
>>><sip:915125656553@backup.ccsi.com>;tag=E8186070-487.
>>>
>>>
>>>
>>>
>>>> .Date: Tue, 15 Feb 2000 01:38:28 GMT..Call-ID:
>>>>9fef06800312431fbaa33d389f7d
>>>> 3ac7@192.168.1.101..Server:
>>>>Cisco-SIPGateway/IOS-12.x..CSeq: 1
>>>>INVITE..Allo
>>>> w-Events: telephone-event..Content-Length: 0....
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
>>
>>
>>
>>
>>>>C.M. Rahman Jr.
>>>>CTO
>>>>CCNP, MCSE Security "Secure your self by
>>>>
>>>>
>>>>
>>>>
>>>securing
>>>
>>>
>>>
>>>
>>>>your System"
>>>>CompTI Security Plus Certified
>>>>CCS Internet
>>>>http://www.ccsi.com
>>>>13704 Research Blvd. Building O-Suite 4
>>>>Austin, TX 78750
>>>>Tel: 512-257-2274 Ex: 115
>>>>
>>>>
>>>>-----Original Message-----
>>>>From: Stephen Kingham
>>>>[mailto:Stephen.Kingham@aarnet.edu.au]
>>>>Sent: Thursday, June 24, 2004 11:56 PM
>>>>To: CM Rahman
>>>>Cc: serusers(a)lists.iptel.org
>>>>Subject: Re: [Serusers] as5400 and ser
>>>>
>>>>Hi
>>>>
>>>>Along with several other we are putting together a
>>>>SER implementation
>>>>Tutorial for the R&E sector.
>>>>
>>>>We have a page up the the AS5300 and it may help
>>>>you, also if anyone is
>>>>interested in reviewing it?
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipwork
s
>>
>>
>h
>
>
>>
>>
>>
>>
>>>>op/uas/ciscoas5300.html
>>>>
>>>>Regards
>>>>
>>>>Stephen
>>>>
>>>>CM Rahman wrote:
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>>Anybody here using cisco as5400 for PSTN
>>>>>
>>>>>
>>>>>
>>>>>
>>>>termination? I am having some
>>>>
>>>>
>>>>
>>>>
>>>>>problem with call routing. If there are such
>>>>>
>>>>>
>>>>>
>>>>>
>>>person
>>>
>>>
>>>
>>>
>>>>will to help,
>>>>please
>>>>
>>>>
>>>>
>>>>
>>=== message truncated ===
>>
>>
>>
>>
>>__________________________________
>>Do you Yahoo!?
>>Read only the mail you want - Yahoo! Mail SpamGuard.
>>http://promotions.yahoo.com/new_mail
>>
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>>
>>
>>
>>
>
>
>
I moved ser from inside my office to a public IP address. When I use
X-Lite it connects to ser, tries to login anonymously and ser sends back the
Unauthorized packet. When it was in the office X-Lite would then send the
name/password and login. It doesn't do that now.
I'm using a LinkSys router. When the packet comes back to the LinkSys
router from ser it hits the LinkSys at port 5060. I don't believe that's
correct but I can't figure out how to correct it.
I enabled nathelper but am not sure what to do next.
Also, where is the documentation for the nat_uac_test? I can't find a
single refererence to that in my source code anywhere.
Bill
Actually I have a Lucent Excel switch which is connected to the cisco
as5400 via T1 Pri. Anybody here using Excel switch with a cisco ?
Right now, when ever I do debug q931 I get this below and it hangs until
my messenger times out and it disconnects. It should answer and give me
voice prompt. Anybody have deal with same scenario as mine?
*Feb 18 15:40:20.142: ISDN Se7/0:3:23 Q931: Applying typeplan for
sw-type 0xD is 0x2 0x1, Called num 5122200090
*Feb 18 15:40:20.142: ISDN Se7/0:3:23 Q931: TX -> SETUP pd = 8 callref
= 0x005F
Bearer Capability i = 0x8090A2
Standard = CCITT
Transer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Called Party Number i = 0xA1, '5122200090'
Plan:ISDN, Type:National
*Feb 18 15:40:20.158: ISDN Se7/0:3:23 Q931: RX <- CALL_PROC pd = 8
callref = 0x805F
Channel ID i = 0xA98381
Exclusive, Channel 1
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
C.M. Rahman Jr.
CTO
CCNP, MCSE Security "Secure your self by securing your System"
CompTI Security Plus Certified
CCS Internet
http://www.ccsi.com
13704 Research Blvd. Building O-Suite 4
Austin, TX 78750
Tel: 512-257-2274 Ex: 115
-----Original Message-----
From: Stephen Kingham [mailto:Stephen.Kingham@aarnet.edu.au]
Sent: Monday, June 28, 2004 5:50 AM
To: CM Rahman
Cc: Richard; serusers(a)lists.iptel.org
Subject: Re: [Serusers] as5400 and ser
CM Rahman wrote:
>I am sorry, I didn't show how put the pot in my last email, here it is,
>
>dial-peer voice 150 voip
> description CCSi voip phone
> destination-pattern 9T
> progress_ind setup enable 3
> session protocol sipv2
> session target ipv4:216.236.160.11
> codec g723r53
>
>
>
>Answer to your question, without putting "isdn protocol-emulate
network"
>I wasn't able to get PRI Layer 2 up.
>
>
Yes. ISDN has a network side and a user side so that the layer 2
protocol Q921/lapb will work.
Most PABX want to be the user side.
>Any other suggestion?
>
>
yes you have to have a pots dialpeer, the Cisco VoIP gateway requires at
least one, I think maybe one for each E1 port.
Take a look at the template I have posted here:
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworksh
op/uas/ciscoVoIPGateways/as5300-12.3-6b-sip.txt
>
>&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
>C.M. Rahman Jr.
>CTO
>CCNP, MCSE Security "Secure your self by securing your System"
>CompTI Security Plus Certified
>CCS Internet
>http://www.ccsi.com
>13704 Research Blvd. Building O-Suite 4
>Austin, TX 78750
>Tel: 512-257-2274 Ex: 115
>
>-----Original Message-----
>From: Richard [mailto:mypop3mail@yahoo.com]
>Sent: Friday, June 25, 2004 4:11 PM
>To: CM Rahman; serusers(a)lists.iptel.org
>Subject: RE: [Serusers] as5400 and ser
>
>Don't know why you have the following two lines,
>isdn protocol-emulate network
>isdn incoming-voice modem
>
>Also you probably need a pots dial-peer...
>
>Cisco web site has some configuration samples.
>
>--- CM Rahman <cmrahman(a)ccsi.com> wrote:
>
>
>>Once I send a call via messenger, I don't hear
>>anything other side. But
>>after a while it disconnect.
>>
>>Here are the cisco config
>>
>>******************************
>>controller T1 7/0:3
>> framing esf
>> pri-group timeslots 1-24
>> description Prism Test
>>
>>***************************************
>>interface Serial7/0:3:23
>> no ip address
>> isdn switch-type primary-ni
>> isdn protocol-emulate network
>> isdn incoming-voice modem
>> isdn T310 180000
>> no cdp enable
>>!***************************************
>>
>>dial-peer voice 150 voip
>> description CCSi voip phone
>> destination-pattern 9T
>> session protocol sipv2
>> session target ipv4:216.236.160.11
>> codec g723r53
>>
>>*****************************************
>>
>>
>>
>>
>>*Feb 15 16:18:09.720: ISDN Se7/0:3:23 Q931: Applying
>>typeplan for
>>sw-type 0xD is 0x2 0x1, Called num 5122200090
>>*Feb 15 16:18:09.720: ISDN Se7/0:3:23 Q931: TX ->
>>SETUP pd = 8 callref
>>= 0x002E
>> Bearer Capability i = 0x8090A2
>> Standard = CCITT
>> Transer Capability = Speech
>> Transfer Mode = Circuit
>> Transfer Rate = 64 kbit/s
>> Channel ID i = 0xA98381
>> Exclusive, Channel 1
>> Called Party Number i = 0xA1, '5122200090'
>> Plan:ISDN, Type:National
>>*Feb 15 16:18:09.732: ISDN Se7/0:3:23 Q931: RX <-
>>CALL_PROC pd = 8
>>callref = 0x802E
>> Channel ID i = 0xA98381
>> Exclusive, Channel 1
>>*Feb 15 16:20:17.967: ISDN Se7/0:3:23 Q931: TX ->
>>DISCONNECT pd = 8
>>callref = 0x002E
>> Cause i = 0x8290 - Normal call clearing
>>*Feb 15 16:20:17.995: ISDN Se7/0:3:23 Q931: RX <-
>>RELEASE pd = 8
>>callref = 0x802E
>>*Feb 15 16:20:17.995: ISDN Se7/0:3:23 Q931: TX ->
>>RELEASE_COMP pd = 8
>>callref = 0x002E
>>
>>
>>
>>
>&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
>
>
>>C.M. Rahman Jr.
>>CCNP, MCSE Security "Secure your self by securing
>>your System"
>>CompTI Security Plus Certified
>>CCS Internet
>>http://www.ccsi.com
>>13704 Research Blvd. Building O-Suite 4
>>Austin, TX 78750
>>Tel: 512-257-2274 Ex: 115
>>
>>-----Original Message-----
>>From: serusers-bounces(a)lists.iptel.org
>>[mailto:serusers-bounces@lists.iptel.org] On
>>Behalf Of Richard
>>Sent: Friday, June 25, 2004 3:27 AM
>>To: serusers(a)lists.iptel.org
>>Subject: RE: [Serusers] as5400 and ser
>>
>>If you check this page,
>>
>>
>>
>>
>http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_
g
>
>
>>uide_chapter09186a00800eadfa.html
>>
>>PSTN error "63 Service or option unavailable" is
>>mapped to sip error "503 Service or option
>>unavailable" which is in the header of the message.
>>
>>Also the page shows why IP phone or PSTN generates
>>this and how proxy is supposed to do with it. Quote,
>>"The SIP gateway generates this response if it is
>>unable to process the request due to an overload or
>>maintenance problem. Upon receiving this response,
>>the
>>gateway initiates a graceful call disconnect and
>>clears the call. "
>>
>>Look like a pstn config issue. Use "debug isdn
>>q931",
>>"debug isdn q921" and "term mon" for further
>>debuging.
>>
>>Cheers,
>>Richard
>>
>>--- CM Rahman <cmrahman(a)ccsi.com> wrote:
>>
>>
>>>Looking through your cisco config file, I am
>>>guessing your E1 are not
>>>Pri. Ami I correct? I am dealing with a
>>>
>>>
>>channelized
>>
>>
>>>DS3 with T1 Pri. I
>>>will also share my config file after I can get the
>>>call routed.
>>>Currently I am getting this below. My
>>>
>>>
>>understanding
>>
>>
>>>is there is
>>>something wrong in the call going from cisco to
>>>
>>>
>>Pri
>>
>>
>>>trunk. Anybody can
>>>give me some clue, that will be great.
>>>
>>>
>>>
>>>146.82.136.218:5060 -> 216.236.160.11:5060
>>> SIP/2.0 503 Service Unavailable..Via:
>>>
>>>
>>SIP/2.0/UDP
>>
>>
>>>216.236.160.11;branch=z9h
>>> G4bKc513.1c338976.0,SIP/2.0/UDP
>>>65.70.207.66:8675..From:
>>>"pappusip(a)backup.c
>>> csi.com"
>>>
>>>
>>>
><sip:pappusip@backup.ccsi.com>;tag=c270cb2a9ab14343b72218adb808612
>
>
>>> 4;epid=c91b05026b..To:
>>>
>>>
>>>
>><sip:915125656553@backup.ccsi.com>;tag=E8186070-487.
>>
>>
>>> .Date: Tue, 15 Feb 2000 01:38:28 GMT..Call-ID:
>>>9fef06800312431fbaa33d389f7d
>>> 3ac7@192.168.1.101..Server:
>>>Cisco-SIPGateway/IOS-12.x..CSeq: 1
>>>INVITE..Allo
>>> w-Events: telephone-event..Content-Length: 0....
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
>
>
>>>C.M. Rahman Jr.
>>>CTO
>>>CCNP, MCSE Security "Secure your self by
>>>
>>>
>>securing
>>
>>
>>>your System"
>>>CompTI Security Plus Certified
>>>CCS Internet
>>>http://www.ccsi.com
>>>13704 Research Blvd. Building O-Suite 4
>>>Austin, TX 78750
>>>Tel: 512-257-2274 Ex: 115
>>>
>>>
>>>-----Original Message-----
>>>From: Stephen Kingham
>>>[mailto:Stephen.Kingham@aarnet.edu.au]
>>>Sent: Thursday, June 24, 2004 11:56 PM
>>>To: CM Rahman
>>>Cc: serusers(a)lists.iptel.org
>>>Subject: Re: [Serusers] as5400 and ser
>>>
>>>Hi
>>>
>>>Along with several other we are putting together a
>>>SER implementation
>>>Tutorial for the R&E sector.
>>>
>>>We have a page up the the AS5300 and it may help
>>>you, also if anyone is
>>>interested in reviewing it?
>>>
>>>
>>>
>>>
>http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworks
h
>
>
>>>op/uas/ciscoas5300.html
>>>
>>>Regards
>>>
>>>Stephen
>>>
>>>CM Rahman wrote:
>>>
>>>
>>>
>>>>Anybody here using cisco as5400 for PSTN
>>>>
>>>>
>>>termination? I am having some
>>>
>>>
>>>>problem with call routing. If there are such
>>>>
>>>>
>>person
>>
>>
>>>will to help,
>>>please
>>>
>>>
>=== message truncated ===
>
>
>
>
>__________________________________
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>
>
>_______________________________________________
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>
>
--
Stephen Kingham, MIT, BSc, E&C Cert
Project Manager and Consulting Engineer
mailto:Stephen.Kingham@aarnet.edu.au
Telephone +61 2 6222 3575 (office)
+61 419 417 471 (mobile)
Voice and Video over IP
for The Australian Academic Research Network (AARNet) and
http://www.aarnet.edu.au
Hello Everyone,
I am trying to change the URI within the ser.cfg using the replace
command. Has anyone done this before? What I need to do is strip some
of the info after the "@" and forward the message back out. So, I can't
necessarily use the rewritehostport (or rewritehost, rewriteport)
commands. Ay help would be greatly appreciated..
Simple Example
INVITE sip:mark@sip.abc.def.com:5060
Want to change to
INVITE sip:mark@sip.def.com:5060
Thanks,
Mark
Hi,
The functions;
setflag()
restflag()
isflagset()
Do these functions set a flag identified by the numeric argument, or do
they set one specific flag to the value of the argument?
I'm sure it is the former, but the docs are not that clear to me on this.
Also, if I update the docs, where is the best place for me to submit
changes to? Shall I just email the sgml diffs to the serdev list?
All the best,
-Jev
Hey can any one answer this
I want SER to act as a sip proxy for TCP as well as UDP.
UA -------tcp/udp------- SER ----tcp------- LCS (TCP)
|
|_____udp____ Asterisk (UDP)
Also the SER should be able to take a udp request and change it to tcp
and send it out or vice versa
Can I do this ????
I am sure one of you have faced this problem and have doen something
Please share with me your thoughts.
thanks
Dipen K Gala
Fidelity Investments System Company - Telecom
Phone: (617) 563-2729
Email: Dipen.Gala(a)FMR.com
