Hi,
I want to launch 30 simultaneous calls from one user agent towards a VoIP Gateway. I would like to know if you know any software to perform this simultaneous calls.
Marcos
Hi!
Regarding IM within SIP, there are two modes: paging and session.
The first is a kind of sms service, where users sends *few* messages(if you are not a teenager... ;) ) . This is provided by the standard SIP request MESSAGE.
The session mode provides the typical chat scenario, where users are typing *constantly* message and they are exchanged in (soft) real time. This is provided by the MSRP protocol defined by the IETF's group SIMPLE. In that case, the MSRP session parameters are carried in the SDP body of the INVITE. Once the Offer/Answer is completed, both end points can send messages via the MSRP session.
For both modes, SIP proxies (SER) do not require extended capabilities because in both cases SIP requests (MESSAGE for paging and INVITE for session) are proxied normally and finally arrive to the end points. The main difference lies in the UA: the paging mode is widely supported because almost all UA supports MESSAGE request, while the session mode is not yet so extended because MSRP is quite new.
Although Messenger started (>4.6 I think) with plain SIP supporting only MESSAGE requests, it needed a chat session mode to be implemented and since MSRP was not still finished, microsoft decided to develop its own protocol (strange, isn't it?). That is why new versions of microsoft messenger are not compatible with standard SIP UA.
Once MSRP is extended (there's an open source implementation in www.sipfoundry.org), standard SIP UA (excluding microsfot thing) will be fully interoperable both in paging mode (MESSAGE) and session mode (MSRP).
My 0.02,
Samuel.
Unclassified.
>>> "Victor Huertas Garcia" <vhuertas(a)hotmail.com> 07/01/05 12:40PM >>>
Hi all,
I am testing the Instant Message Service with the SER v0.8.14 and I see that
it routes them correctly towards the destination user agent. However, what I
have seen is a tremendous imcompatibility between user agents of different
vendors.
Here just a bit of this:
IM between two Windows Messenger 4.7.
What I see it that the very first MESSAGE in the conversation is sent
towards the SER and it routes it perfectly but the subsequent MESSAGES are
exchange between both Messengers directly!!! It is weird... but this is how
it is working.
IM between Window Messenger 4.7 and EyeBeam v1.1
The EyeBeam always send the MESSAGES towards the SIP proxy and messenger
receive them with no problem. However, when the messenger tries to send a
MESSAGE in the conversation it even doesn't send the packet (I tried to
capture it with Ethereal and no packet was sent at all) and subsequently
tell you that the text message could not be delivered to the destination.
IM between EyeBeam v1.1
No problem at all.
IM involving Windows Messenger 5.1
This version of messenger sends an INVITE to the destination before issuing
the MESSAGE. With EyeBeam is a total disaster and the comunication is not
possible.
My conclusion is that the SER has not implication in such incompatibility at
all and that is a matter of interoperability among vendors and different
implementations of SIMPLE. Do you agree?
Thanks in advance
Victor
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http://lists.iptel.org/mailman/listinfo/serusers
Hi there!!!... Iam newy in SER mailing list, and I have a problem with my
SER installation that I saw that other members have before, but I can't find
a solution to the issue.
I have installed the SER (v0.8.14), all works fine until I add MySQL support
to get persistent users. after that, when I run "serctl start" I get this
error:
----------------------------------------------------------------
Starting SER : cat: /var/run/ser.pid: No such file or directory
started pid()
----------------------------------------------------------------
I have made some test to troubleshot this issue, for exsaplme: remplacing
DIR=... with the full path to SER executable in "serctl"... and also I
increase the debug level in "ser.cfg" to 7, and I get this output:
----------------------------------------------------------------
MyHost:/etc/ser# ser -c
0(2577) loading module /usr/lib/ser/modules/mysql.so
0(2577) loading module /usr/lib/ser/modules/sl.so
0(2577) loading module /usr/lib/ser/modules/tm.so
0(2577) loading module /usr/lib/ser/modules/rr.so
0(2577) loading module /usr/lib/ser/modules/maxfwd.so
0(2577) loading module /usr/lib/ser/modules/usrloc.so
0(2577) loading module /usr/lib/ser/modules/registrar.so
0(2577) loading module /usr/lib/ser/modules/auth.so
0(2577) loading module /usr/lib/ser/modules/auth_db.so
0(2577) set_mod_param_regex: usrloc matches module usrloc
0(2577) set_mod_param_regex: found <db_mode> in module usrloc
[/usr/lib/ser/modules/usrloc.so]
0(2577) set_mod_param_regex: auth_db matches module auth_db
0(2577) set_mod_param_regex: found <calculate_ha1> in module auth_db
[/usr/lib/ser/modules/auth_db.so]
0(2577) set_mod_param_regex: auth_db matches module auth_db
0(2577) set_mod_param_regex: found <password_column> in module auth_db
[/usr/lib/ser/modules/auth_db.so]
0(2577) set_mod_param_regex: rr matches module rr
0(2577) set_mod_param_regex: found <enable_full_lr> in module rr
[/usr/lib/ser/modules/rr.so]
0(2577) find_export: found <mf_process_maxfwd_header> in module
maxfwd_module [/usr/lib/ser/modules/maxfwd.so]
0(2577) find_export: found <sl_send_reply> in module sl_module
[/usr/lib/ser/modules/sl.so]
0(2577) find_export: found <sl_send_reply> in module sl_module
[/usr/lib/ser/modules/sl.so]
0(2577) find_export: found <record_route> in module rr
[/usr/lib/ser/modules/rr.so]
0(2577) find_export: found <loose_route> in module rr
[/usr/lib/ser/modules/rr.so]
0(2577) find_export: found <t_relay> in module tm
[/usr/lib/ser/modules/tm.so]
0(2577) find_export: found <www_authorize> in module auth_db
[/usr/lib/ser/modules/auth_db.so]
0(2577) find_export: found <www_challenge> in module auth
[/usr/lib/ser/modules/auth.so]
0(2577) find_export: found <save> in module registrar
[/usr/lib/ser/modules/registrar.so]
0(2577) find_export: found <lookup> in module registrar
[/usr/lib/ser/modules/registrar.so]
0(2577) find_export: found <sl_send_reply> in module sl_module
[/usr/lib/ser/modules/sl.so]
0(2577) find_export: found <t_relay> in module tm
[/usr/lib/ser/modules/tm.so]
0(2577) find_export: found <sl_reply_error> in module sl_module
[/usr/lib/ser/modules/sl.so]
0(2577) routing table 0:
...blah
...blah
...blah
...blah
WARNING: could not rev. resolve 192.168.0.4
Listening on
127.0.0.1 [127.0.0.1]:5060
192.168.0.4 [192.168.0.4]:5060
Aliases: MyHost:5060 localhost:5060 localhost.localdomain:5060
config file ok, exiting...
0(2577) DEBUG: tm_shutdown : start
0(2577) DEBUG: tm_shutdown : empting hash table
0(2577) DEBUG: tm_shutdown: releasing timers
0(2577) DEBUG: tm_shutdown : removing semaphores
0(2577) DEBUG: tm_shutdown : done
0(2577) shm_mem_destroy
0(2577) destroying the shared memory lock
----------------------------------------------------------------
I couldn't find what is wrong in my configuration...
BTW, I can add, remove and edit users in MySQL Database using "serctl"
subscribers commands (that means that my SER machine is talking with MySQL
server as in expected)
Is there a way to have multiple sip addresses for one account in
serweb? For example, if a group or business wanted to have one
login/account to view the activity/preferences for all of their sip
addresses.
Thanks!
- Daryl
Hi, Steve.....
I want to ask about dial-peers u provide in ur
website(http://mit.edu/sip/sip.edu/ciscoGW.html)...
1. dial-peer voice 680010 voip
description Only peer for inbound to
SIP Proxy 215-746-8001:8009 extensions
huntstop
preference 2
destination-pattern 6800[1-9]
progress_ind setup enable 3
voice-class codec 1
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
what is 680010?????? is that ur phone number????
something related to phone number???
2. dial-peer voice 61 pots
description Only peer for outbound 5-digit 746
campus calls
translation-profile outgoing Prefix
preference 3
destination-pattern 6....
direct-inward-dial
port 1/0:23
prefix 215746
why do u use 61???? something related to ur pone
number????
3. could u give me some other example configuration,
bcoz I dun use PABX or analog router here.. I plug in
my Telephone line direct to VIC2FXO card...
so I Wish that my SIP client can call to PSTN
client...
my Telephone number that I plug to cisco router is
62(761) 53808 , 62 is country code, 761 is area kode,
53808 is my phone number...
__________________________________
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Hello,
I do have the following in ser.cfg:
------------------------
failure_route[1] {
log(1, "Entering failure_route(1)");
revert_uri();
log(1, "reverted");
rewritehostport("xxx.xxx.xxx.xxx:5060");
log(1, "rewritten");
append_branch();
log(1, "appended");
t_relay_to_udp("xxx.xxx.xxx.xxx", "5060");
log(1, "relayed");
log(1, "Exiting failure_route(1)");
break;
}
------------------------
It is called from there:
-----------------------
route[1] {
if (method=="INVITE") {
t_on_failure("1");
}
if (!t_relay()) {
sl_reply_error();
}
else {
log (1, "t_relay() done in route(1)");
}
break;
}
------------------------
Which is called from the end of route{} and also from end of some
route(x).
In the log, I can see SER crashing, after tm's "fr_inv_timer" seconds:
Jun 30 16:40:00 ser /usr/local/sbin/ser[3548]: Entering failure_route
(1) Jun 30 16:40:00 ser /usr/local/sbin/ser[3548]: reverted
Jun 30 16:40:00 ser /usr/local/sbin/ser[3548]: rewritten
Jun 30 16:40:00 ser /usr/local/sbin/ser[3548]: appended
Jun 30 16:40:00 ser /usr/local/sbin/ser[3548]: BUG: qm_free: bad
pointer 0x2a97d717e8 (out of memory block!) - aborting Jun 30
16:40:00 ser /usr/local/sbin/ser[3558]: ERROR: receive_fd: EOF on 14
Jun 30 16:40:00 ser /usr/local/sbin/ser[3525]: child process 3548
exited by a signal 6 Jun 30 16:40:00 ser /usr/local/sbin/ser[3525]:
core was generated Jun 30 16:40:00 ser /usr/local/sbin/ser[3525]:
INFO: terminating due to SIGCHLD
It also do the same if I replace t_relay_to_udp(...) with t_relay().
And also if I move the t_on_failure() in main route() or the calling
route(x).
Is it a known bug or am I doing something wrong?
Any help appreciated!
Best regards
--
# Lol Zimmerli // S y s C o ® // http://www.sysco.ch/
Modularise. Use subroutines.
- The Elements of Programming Style (Kernighan & Plaugher)
hi all,ter around
we use ser with many differents handests, ata and softphones.
with some of them (sipura and Zyxel) the call finished after around 20/30
seconds.
The terminator send us a 200 OK when communication start, but with these
UAc, ser doesn't send back an ACK to the termination.
Any idea of how to solve that problem?
Thanks,
Olivier
Olivier Taylor Geomatics
19 avenue du vivier d'oie
1000 Bruxelles
olivier.taylor(a)geomatics.be
tel:
mobile: +32 2 320 06 16
+32 495 28 33 61
<https://www.plaxo.com/add_me?u=21475503789&v0=1311510&k0=2123456319&v1=1311
511&k1=2037529684> Add me to your address book...
<http://www.plaxo.com/signature> Want a signature like this?
Hi,
Sorry to resubmit on much the same topic as before (Portal for forking
call to preferred end device-sequential ringing) but I was wondering if
anyone had any further ideas on how to associate a particular contact
address with a location. i.e. a user can choose that their preferred
location is their desktop phone and I need to devise some way to know
that a particular contact address is associated with the desktop phone.
I was thinking the user must either enter the device ip addresses
manually so I can search the contact based on IP address (This sucks in
terms of NAT, scalability and losing mobility over networks) OR I could
perhaps stipulate that a user cannot have two of the same device e.g.
cant have two BT100 hardphones and I could search the location table
based on model. I think either of these options aren't great....Which is
why I was wondering if anyone had any better ideas?
Many Thanks,
Aisling.
p.s. I currently retrieve the users contact address and q value via a
web interface using the serctl fifo interface.
-----Original Message-----
From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro]
Sent: 22 June 2005 12:45
To: Aisling
Cc: 'Greger V. Teigre'; jh(a)tutpro.com
Subject: Re: [Serusers]CPL - Portal for forking call to
preferredenddevice-sequential ringing
Hi Aisling,
There are many ways to implement serial forking - like using CPL, LCR or
AVPOPS.
But the main idea (if I getting right from your initial email) is to
allow the user to order his contacts for the same account (like account
userx that have contact_pda_x, contact_PC_x, contact_office_x, etc). So
you have all this contacts in user location, you have several mechanism
for serial forking, but you are missing the link - to get the contacts
from user location and to feed any of the serial forking mechanism. This
is the big problem.
even if you use CPL, you need to place in each location node the
contacts (and not the AOR) of the user, contacts which are dynamically
stored only in user location.
what you are describing below can be possible only if you a user (as
person) has different sip accounts (SIP users) for each of his devices.
Other way I don't see how you can place in the CPL script the "phone1"
and "phone2" addresses.
and just to answer you to the CPL- related questions (from
implementation point of view):
- each user can have only one script - if you load a new script, the
previous one will be lost (overwritten). If you can different services
via CPL (like screening and no-answer), you need to mix them in the same
script during provisioning.
- if you want to delete a CPL script via FIFO, use the REMOVE_CPL
command:
serctl fifo REMOVE_CPL user@domain
regards,
bogdan
Aisling wrote:
>Hello,
>
>Many thanks for the ideas so far.
>I looked at CPL Greger and I think that provides a very simple solution
>to this - a simple Call forward on no answer script.
>
>I have an included an example that I came across below. If a user wants
>to modify the order of the devices a call should be sent to, then I
>simply have to retrieve the information from the user via the web
>interface and provision a new cpl script. I think this solves the
>problem - Does anyone foresee any problems with this or think it has
>disadvantages?
>
>I do have two minor questions if I am to go ahead with this direction
>though:
>1) If a particular user already has a cpl script e.g. a call screening
>script uploaded to the database and they then upload this forward on no
>answer script, will it overwrite the original script? i.e. can there
>only be one cpl script per user?
>
>2)How can a cpl script be "undone" or deleted? Must it be overwritten
or
>is there a way of simply removing it(without using mysql commands)?
>
>Example CPL: Call Forward on No Answer
>
><?xml version="1.0">
>
><cpl>
> <subaction id="phone2">
> <location url="sip:2000@phone2">
> <proxy />
> </location>
> </subaction>
>
> <incoming>
> <location url="sip:2000@phone1">
> <proxy timeout="8">
> <noanswer>
> <sub ref="phone2"/>
> </noanswer>
> </proxy>
> </location>
> </incoming>
></cpl>
>
>Many Thanks,
>Aisling.
>
>-----Original Message-----
>From: Greger V. Teigre [mailto:greger@teigre.com]
>Sent: 22 June 2005 05:42
>To: Aisling O'Driscoll; jh(a)tutpro.com
>Cc: samuel.osorio(a)nl.thalesgroup.com; ashling.odriscoll(a)cit.ie;
>serusers(a)lists.iptel.org
>Subject: Re: [Serusers] Portal for forking call to
>preferredenddevice-sequential ringing
>
>Aisling O'Driscoll wrote:
>
>
>>Ok, just to recap - cos Im getting a little bit confused ;)
>>
>>I have two choices(?)
>>
>>1. Somehow invoke sipsak to configure permanent addresses with a
>>particular q value.
>>2. Develop a FIFO method to change q value.
>>
>>
>
>Yes. Except that I don't know if anybody verified that q value cannot
be
>
>changed with today's FIFO. I just asked the question...
>
>
>
>>Am I correct in thinking directly modifying the usrloc table in the
>>database is out of the question because the changes cant be updated
>>except by SER itself in which case a reboot would required - Correct?
>>
>>
>
>Correct. "SER itself" is here either serctl, sipsak or FIFO.
>
>
>
>>Also lcr module (load_contacts() etc) isnt suitable for per user
>>configurable sequential forking?
>>
>>
>
>I don't know about that. There is definitely a q-value based
>functionality
>there.
>
>
>
>>Have others tried to implement similar functionality or is it usually
>>a generic site wide sequential forking policy?
>>
>>
>
>I think using CPL could be an option. Have you looked at CPL and the
cpl
>
>module?
>g-)
>
>
>
>>Many thanks for the opinions and help so far.
>>Aisling.
>>
>>
>>
>>>---- Original Message ----
>>>From: jh(a)tutpro.com
>>>To: greger(a)teigre.com
>>>Subject: Re: [Serusers] Portal for forking call to
>>>preferredenddevice-sequential ringing
>>>Date: Tue, 21 Jun 2005 21:41:54 +0300
>>>
>>>
>>>
>>>>Greger V. Teigre writes:
>>>>
>>>>
>>>>
>>>>>:-D Yes, that is true. But I didn't know that you could change
>>>>>
>>>>>
>>>q-value in
>>>
>>>
>>>>>FIFO?
>>>>>
>>>>>
>>>>if that is not possible, then you can always use sipsak to install
>>>>"permanent" registrations with a given q value. sipsak has an
>>>>additional advantage over fifo in that you can apply permissions
>>>>
>>>>
>>>check
>>>
>>>
>>>>to sipsak registration, but not to fifo registration.
>>>>
>>>>-- juha
>>>>
>>>>
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Hi!
The 0.9.4 source tarballs at http://openser.org/pub/openser/0.9.4/src/
still have the postgres-usrloc bug. (I knew I should have used CVS)
I suggest that they will be updated automatically, or removed.
regards,
klaus