[OpenSIPS-Users] How to force opensips add port field in via header?
Hi everyone, I'm using opensips 1.6. When I send sip msg by call forward function, opensips add Via header in sips message like "Via: SIP/2.0/UDP 10.37.143.6;branch=z9hG4bKSS1vtZ6ZQ98rp." THE QUESTION IS: How can I make opensips add the port field to the VIA header even my sip port is 5060? This mean, the VIA header should be "Via: SIP/2.0/UDP 10.37.143.6*:5060*;branch=z9hG4bKSS1vtZ6ZQ98rp." Does someone can give me some advice about this question? Thank a lot! -- Lei.Tang lei.tl...@gmail.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)
Hi guys, I've got a problem with "sporadic" one-way audio, using latest stable Opensips 1.4 and Mediaproxy 2. The problem occurs at re-invites, thou not every time. When it does happen, the media port is changed towards the callee. Example: First invite: INVITE Trunk:40518 --> Proxy:50358 INVITE Proxy:58928 --> UA:5206 Re-invite: INVITE Trunk:40518 --> Proxy:50358 INVITE Proxy:40518 --> UA:5206 Notice how the Proxy port on the UA side (callee local) changed to the same port that Trunk uses (caller remote). One way audio now occurs and naturally the call is soon hung up. Mediaproxy now dumps some useful statistics showing: 'callee_remote': 'UA:5206' 'caller_remote': 'TRUNK:40518' 'callee_local': 'PROXY:58928' 'caller_local': 'PROXY:50358' Notice here how the callee_local is the original port used. Where should I start looking to be able to solve this problem? Is it most likely my config, something weird with the re-invite or have I stumbled upon a bug? Thankful for every and all suggestions, as I lay sleepless at night thinking about this! :-P Cheers, Magnus ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to force opensips add port field in via header?
El Martes, 2 de Febrero de 2010, Lei Tang escribió: > Hi everyone, I'm using opensips 1.6. > When I send sip msg by call forward function, opensips add Via header in > sips message like "Via: SIP/2.0/UDP > 10.37.143.6;branch=z9hG4bKSS1vtZ6ZQ98rp." >THE QUESTION IS: How can I make opensips add the port field to the VIA > header even my sip port is 5060? This mean, the VIA header should be > "Via: SIP/2.0/UDP 10.37.143.6*:5060*;branch=z9hG4bKSS1vtZ6ZQ98rp." >Does someone can give me some advice about this question? Thank a lot! > There is a core parameter for that, I can not remember now the name but you can find it in the wiki (cookbook section). -- Iñaki Baz Castillo ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)
Hi, El 02/02/10 11:54, Magnus Burman escribió: > Hi guys, > > I've got a problem with "sporadic" one-way audio, using latest stable > Opensips 1.4 and Mediaproxy 2. > > The problem occurs at re-invites, thou not every time. When it does > happen, the media port is changed towards the callee. Example: > > First invite: > INVITE Trunk:40518 --> Proxy:50358 > INVITE Proxy:58928 --> UA:5206 > > Re-invite: > INVITE Trunk:40518 --> Proxy:50358 > INVITE Proxy:40518 --> UA:5206 > > Notice how the Proxy port on the UA side (callee local) changed to the > same port that Trunk uses (caller remote). > > One way audio now occurs and naturally the call is soon hung up. > Mediaproxy now dumps some useful statistics showing: > > 'callee_remote': 'UA:5206' > 'caller_remote': 'TRUNK:40518' > 'callee_local': 'PROXY:58928' > 'caller_local': 'PROXY:50358' > > Notice here how the callee_local is the original port used. > > Where should I start looking to be able to solve this problem? Is it > most likely my config, something weird with the re-invite or have I > stumbled upon a bug? > > Thankful for every and all suggestions, as I lay sleepless at night > thinking about this! :-P > It would be nice to have a full SIP trace and the syslog output of MediaProxy to check if it's doing something wrong. Also, which function are you using for starting it? Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Error in module permission with db_text
Hi, Bogdan.
I have got one more error with module db_text.
loadmodule "avpops.so"
modparam("avpops","db_url","text:///etc/opensips/dbtext/")
modparam("avpops","avp_table","usr_preferences")
modparam("avpops","use_domain",0)
modparam("avpops","uuid_column","uuid")
modparam("avpops","username_column","username")
modparam("avpops","domain_column","domain")
modparam("avpops","attribute_column","attribute")
modparam("avpops","type_column","type")
modparam("avpops","value_column","value")
...
avp_db_load("mediaproxy-RU","$avp(s:mediaproxy_ip)")
...
usr_preferences:
id(int,auto) uuid(str) username(str) domain(str) attribute(str) type(int)
value(str) last_modified(double,null)
10:mediaproxy-DEFAULT:mediaproxy-DEFAULT:domain:mediaproxy_ip:0:value:0
20:mediaproxy-RU:mediaproxy-RU:domain:mediaproxy_ip:0:value:0
And I see in log:
Feb 2 12:03:51 dev-sip /usr/sbin/opensips[19536]:
DBG:db_text:dbt_load_file: loading file
[/etc/opensips/dbtext//usr_preferences]
Feb 2 12:03:51 dev-sip /usr/sbin/opensips[19536]: ERROR:db_text:dbt_query:
table does not exist!
Feb 2 12:03:51 dev-sip /usr/sbin/opensips[19536]:
ERROR:avpops:ops_dbload_avps: db_load failed
2010/1/26 Bogdan-Andrei Iancu
> Hi Leonid,
>
> Thanks for testing - I uploaded the fix on the 1.6 branch also.
>
>
> Regards,
> Bogdan
>
> Леонид Наседкин wrote:
>
>> Hi, Bogdan.
>> Its working now. Thanks.
>> 2010/1/26 Bogdan-Andrei Iancu > bog...@voice-system.ro>>
>>
>>
>>Hi Leonid,
>>
>>An official fix is available on SVN trunk (rev 6534). I would
>>really appreciate if you could give it a try and test - if ok, I
>>will do the backport.
>>
>>Thanks and regards,
>>Bogdan
>>
>>
>>Bogdan-Andrei Iancu wrote:
>>
>>Ok, I will investigate to come up with an official fix.
>>
>>Thanks and regards,
>>Bogdan
>>
>>Леонид Наседкин wrote:
>>
>>Hi Bogdan
>>Thank you. Its working now.
>>
>>2010/1/15 Bogdan-Andrei Iancu >
>>>>>
>>
>> Hi Leonid,
>>
>> Looks like there is a compatibility bug between
>>permission and
>> db_text modules when comes to DB data typesGive me
>>couple of
>> days to sort this out.
>>
>> In the mean while, if you want to use db_text for
>>permissions,
>> please use the attached patch.
>>
>> Thanks and regards,
>> Bogdan
>>
>>
>> Леонид Наседкин wrote:
>>
>> Hi there.
>> I'm trying to use permission module with db_text,
>>and it's not
>> working, and I can't understand what's wrong.
>> Opensips 1.6.1 svnrevision: 2:6509
>>
>> In opensips.cfg:
>> loadmodule "db_text.so"
>> modparam("db_text", "db_mode", 0)
>> loadmodule "permissions.so"
>> modparam("permissions","db_url",
>>"text:///etc/opensips/dbtext")
>>
>> In /etc/opensips/dbtext/address:
>> id(int,auto) grp(int) ip(str) mask(int) port(int)
>>proto(str)
>> pattern(str,null) context_info(str,null)
>> 10:1:10.100.0.0:23:5060:udp::
>> 20:1:10.110.0.0:23:5060:udp::
>> 30:1:10.120.0.0:23:5060:udp::
>>
>> LOG:
>>
>> DBG:core:init_mod: initializing module permissions
>> DBG:permissions:mod_init: initializing...
>> WARNING:permissions:parse_config_file: file not found:
>> /etc/opensips/permissions.allow
>> WARNING:permissions:mod_init: default allow file
>> (/etc/opensips/permissions.allow) not found =>
>>empty rule set
>> WARNING:permissions:parse_config_file: file not found:
>> /etc/opensips/permissions.deny
>> WARNING:permissions:mod_init: default deny file
>> (/etc/opensips/permissions.deny) not found => empty
>>rule set
>> DBG:core:find_mod_export: found in module
>> db_text [/usr/lib/opensips/modules/]
>> DBG:core:db_bind_mod: using db bind api for db_text
>> INFO:db_text:dbt_init: using database at:
>>/etc/opensips/dbtext/
>> DBG:db_text:dbt_cache_get_db: looking for db
>> /etc/opensips/dbtext/!
>> DBG:db_text:dbt_cache_get_db: new db!
>> DBG:db_text:dbt_load_file: request for table [version]
>> DBG:db_text:dbt_load_file: db is
>>[/etc/opensips/dbtext/]
>> DBG:db_text:dbt_load_file: loading file
>>
Re: [OpenSIPS-Users] How to force opensips add port field in via header?
Hi Iñaki, actually there is no param to force adding port to the local VIA when the port is the SIP default one (just checked the code). Lei, if it really required to get such a via (even if not conforming to RFC), I can send you a small patch. Regards, Bogdan Iñaki Baz Castillo wrote: > El Martes, 2 de Febrero de 2010, Lei Tang escribió: > >> Hi everyone, I'm using opensips 1.6. >> When I send sip msg by call forward function, opensips add Via header in >> sips message like "Via: SIP/2.0/UDP >> 10.37.143.6;branch=z9hG4bKSS1vtZ6ZQ98rp." >>THE QUESTION IS: How can I make opensips add the port field to the VIA >> header even my sip port is 5060? This mean, the VIA header should be >> "Via: SIP/2.0/UDP 10.37.143.6*:5060*;branch=z9hG4bKSS1vtZ6ZQ98rp." >>Does someone can give me some advice about this question? Thank a lot! >> >> > > There is a core parameter for that, I can not remember now the name but you > can find it in the wiki (cookbook section). > > -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to force opensips add port field in via header?
El Martes, 2 de Febrero de 2010, Bogdan-Andrei Iancu escribió: > Hi Iñaki, > > actually there is no param to force adding port to the local VIA when > the port is the SIP default one (just checked the code). Sure? what about "advertised_port=5060"? :) Some years ago I used it in openser 1.2 to avoid a stupid bug in a stupid gateway (Nortel CS2K, the worst machine in the world). -- Iñaki Baz Castillo ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] high-availability - senario
I deceided not to use a loadbalancer because, the IP address for outbound call would have been different and we want it to be completely transparent for existing Interco. I went with heartbeat + mon I created sip.monitor When the fail over take effect it is not listening on the virtual IP, how do I configure opensips to bind to 0.0.0.0:5060 ? From: users-boun...@lists.opensips.org on behalf of Nigel Daniels Sent: Tue 26/01/2010 11:45 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] high-availability - senario have you considerd using keepalived with vrrp instead ? On Tue, Jan 26, 2010 at 7:03 AM, Julien Chavanton wrote: I am configuring high-availability with heart-beat + LVS + ldirector I do not want to load balance but mostly make sure it will fail over quickly, I have found situation where the in LVS (UDP/TCP) connection never timeout when the remote IP send periodic OPTIONS request for example. I beleive I will have to set very low UDP/TCP time-out, however with such a low time-out I can not load-balance so I will use weighted Round-Robin with very high priority 65535 on the active one and weith 1 on the passive server. Any other suggested way to cluster without load-balancing ? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Nigel Daniels Network & Systems Administrator ConnectAndSell inc. (650)-533-2542 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)
Hi,
I did a wireshark export, 2461 lines, so it's in the pastebin:
http://pastebin.ca/1775584
syslog below:
11.11.X.Y - Trunk subnet
22.22.X.Y - Proxy subnet
98400 - UA DID
X.voip.url.es - UA sip subscriber
Jan 11 22:28:07 sbc1 media-relay[4335]: debug: Got traffic information for
stream: (audio) 11.11.6.144:40518 (RTP: Unknown, RTCP: Unknown) <->
22.22.224.9:50358 <-> 22.22.224.9:58928 <-> Unknown (RTP: 22.22.198.159:5206,
RTCP: Unknown)
Jan 11 22:28:07 sbc1 media-relay[4335]: debug: Got traffic information for
stream: (audio) 11.11.6.144:40518 (RTP: Unknown, RTCP: Unknown) <->
22.22.224.9:50358 <-> 22.22.224.9:58928 <-> Unknown (RTP: 22.22.198.159:5206,
RTCP: 22.22.198.159:5207)
Jan 11 22:28:07 sbc1 media-relay[4335]: debug: Got initial answer from
callee for stream: (audio) 11.11.6.144:40518 (RTP: Unknown, RTCP: Unknown)
<-> 22.22.224.9:50358 <-> 22.22.224.9:58928 <-> 22.22.198.159:5206 (RTP:
22.22.198.159:5206, RTCP: 22.22.198.159:5207)
Jan 11 22:28:07 sbc1 media-relay[4335]: debug: Got traffic information for
stream: (audio) 11.11.6.144:40518 (RTP: 11.11.6.144:40518, RTCP: Unknown)
<-> 22.22.224.9:50358 <-> 22.22.224.9:58928 <-> 22.22.198.159:5206 (RTP:
22.22.198.159:5206, RTCP: 22.22.198.159:5207)
Jan 11 22:29:26 sbc1 media-relay[4335]: debug: Got traffic information for
stream: (audio) 11.11.6.144:40518 (RTP: 11.11.6.144:40518, RTCP: Unknown)
<-> 22.22.224.9:50358 <-> 22.22.224.9:58928 <-> 22.22.198.159:5206 (RTP:
22.22.198.159:5206, RTCP: 22.22.198.159:5207)
Jan 11 22:54:20 sbc1 media-dispatcher[4324]: debug: Got statistics:
{'from_tag': 'e-13c4-10b392b-75e19db4-10b392b', 'dialog_id':
'1305:480665684', 'start_time': 1263245287.181, 'timed_out': False,
'call_id': 'a27f94c89e3e13c410b392b13d753bdb84e00e2147f91b8-0266-6714',
'to_tag': 'ICF_197662497-514', 'streams': [{'status': 'closed',
'caller_codec': 'G711a', 'post_dial_delay': 10.76835417749,
'callee_codec': 'G711a', 'start_time': 0, 'caller_bytes': 15733000,
'callee_bytes': 15328600, 'caller_packets': 78665, 'end_time': 1573,
'callee_remote': '22.22.198.159:5206', 'caller_remote': '11.11.6.144:40518',
'media_type': 'audio', 'callee_local': '22.22.224.9:58928', 'timeout_wait':
0, 'caller_local': '22.22.224.9:50358', 'callee_packets': 76643}, {'status':
'rejected', 'caller_codec': 'Unknown', 'post_dial_delay': None,
'callee_codec': 'Unknown', 'start_time': 0, 'caller_bytes': 0,
'callee_bytes': 0, 'caller_packets': 0, 'end_time': 0, 'callee_remote':
'Unknown', 'caller_remote': 'Unknown', 'media_type': 'image',
'callee_local': '22.22.224.9:59676', 'timeout_wait': 0, 'caller_local': '
22.22.224.9:58930', 'callee_packets': 0}], 'duration': 1573, 'to_uri': '
984000...@22.22.224.9:5060', 'from_uri': '34963555...@11.11.0.144:5060',
'callee_ua': 'unknown agent', 'caller_ua': 'CS2000_NGSS/9.0'}
Cheers,
Magnus
2010/2/2 Saúl Ibarra Corretgé
> Hi,
>
> El 02/02/10 11:54, Magnus Burman escribió:
> > Hi guys,
> >
> > I've got a problem with "sporadic" one-way audio, using latest stable
> > Opensips 1.4 and Mediaproxy 2.
> >
> > The problem occurs at re-invites, thou not every time. When it does
> > happen, the media port is changed towards the callee. Example:
> >
> > First invite:
> > INVITE Trunk:40518 --> Proxy:50358
> > INVITE Proxy:58928 --> UA:5206
> >
> > Re-invite:
> > INVITE Trunk:40518 --> Proxy:50358
> > INVITE Proxy:40518 --> UA:5206
> >
> > Notice how the Proxy port on the UA side (callee local) changed to the
> > same port that Trunk uses (caller remote).
> >
> > One way audio now occurs and naturally the call is soon hung up.
> > Mediaproxy now dumps some useful statistics showing:
> >
> > 'callee_remote': 'UA:5206'
> > 'caller_remote': 'TRUNK:40518'
> > 'callee_local': 'PROXY:58928'
> > 'caller_local': 'PROXY:50358'
> >
> > Notice here how the callee_local is the original port used.
> >
> > Where should I start looking to be able to solve this problem? Is it
> > most likely my config, something weird with the re-invite or have I
> > stumbled upon a bug?
> >
> > Thankful for every and all suggestions, as I lay sleepless at night
> > thinking about this! :-P
> >
>
> It would be nice to have a full SIP trace and the syslog output of
> MediaProxy to check if it's doing something wrong. Also, which function
> are you using for starting it?
>
>
>
> Regards,
>
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)
El 02/02/10 13:59, Magnus Burman escribió: > Hi, > > I did a wireshark export, 2461 lines, so it's in the pastebin: > > http://pastebin.ca/1775584 > It's really hard to follow a text wireshark cap, could you do a ngrep capture? ngrep -d any -W byline -T -P '' port 5060 -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Config Suggestions Request (SIP Trunks)
Mike, I have at least one customer on sipXecs. It registers just fine. Granted, there is a bunch of other stuff messed up, but I think that's more my customer's misunderstanding of how to configure it. - Jeff On Feb 2, 2010, at 1:51 AM, Mike O'Connor wrote: > Hi All > > I've a have a couple of customers who are all asking to use sipXecs, > which from my investigations does not support registrations. Instead it > excepts that the ITSP provide unauthenticated trunks for inbound and > outbound calls. > > So my question is what is the recommended way of supporting this ? > > Thanks > Mike > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)
I'll start capturing like that right away, less you want the pcap-file for that specific call? Since the problem only occurs sporadically and at re-invites (ie 20+ minutes into a call) it's a bit of a mess to track down calls that fit the criteria. But I'll get right on it for sure, thanks. Cheers, Magnus 2010/2/2 Saúl Ibarra Corretgé > El 02/02/10 13:59, Magnus Burman escribió: > > Hi, > > > > I did a wireshark export, 2461 lines, so it's in the pastebin: > > > > http://pastebin.ca/1775584 > > > > It's really hard to follow a text wireshark cap, could you do a ngrep > capture? ngrep -d any -W byline -T -P '' port 5060 > > > -- > Saúl Ibarra Corretgé > AG Projects > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)
El Martes, 2 de Febrero de 2010, Magnus Burman escribió: > I'll start capturing like that right away, less you want the pcap-file for > that specific call? Since the problem only occurs sporadically and at > re-invites (ie 20+ minutes into a call) it's a bit of a mess to track down > calls that fit the criteria. But I'll get right on it for sure, thanks. Try this to filter the captured data: http://dev.sipdoc.net/projects/sip-stuff/wiki/Ngrep-SIP -- Iñaki Baz Castillo ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips - using database views
Hi! I'm planning to migrate from fixed database tables to views. But I have a touble to generate the auto increment field - "id". Can opensips operate with tables without the "id" filed? Thanks, Oleg ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips - using database views
Hi! I'm planning to migrate from fixed database tables to views. But I have a touble to generate the auto increment field - "id". Can opensips operate with tables without the "id" filed? Thanks, Oleg ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] high-availability - senario
This was well documented in opensips.cfg listen=udp:0.0.0.0:5060 or for both TCP/UDP listen=0.0.0.0:5060 From: users-boun...@lists.opensips.org on behalf of Julien Chavanton Sent: Tue 02/02/2010 12:47 PM To: OpenSIPS users mailling list; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] high-availability - senario I deceided not to use a loadbalancer because, the IP address for outbound call would have been different and we want it to be completely transparent for existing Interco. I went with heartbeat + mon I created sip.monitor When the fail over take effect it is not listening on the virtual IP, how do I configure opensips to bind to 0.0.0.0:5060 ? From: users-boun...@lists.opensips.org on behalf of Nigel Daniels Sent: Tue 26/01/2010 11:45 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] high-availability - senario have you considerd using keepalived with vrrp instead ? On Tue, Jan 26, 2010 at 7:03 AM, Julien Chavanton wrote: I am configuring high-availability with heart-beat + LVS + ldirector I do not want to load balance but mostly make sure it will fail over quickly, I have found situation where the in LVS (UDP/TCP) connection never timeout when the remote IP send periodic OPTIONS request for example. I beleive I will have to set very low UDP/TCP time-out, however with such a low time-out I can not load-balance so I will use weighted Round-Robin with very high priority 65535 on the active one and weith 1 on the passive server. Any other suggested way to cluster without load-balancing ? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Nigel Daniels Network & Systems Administrator ConnectAndSell inc. (650)-533-2542 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Bulding OpenSIPS v1.6.1-notls on 64 bit machine
Hello, I'm trying to build a fresh copy of OpenSIPS v1.6.1-notls on a 64 bit RH system and experiencing some path issues when running opensipsdbctl create. It seems some scripts exist in /usr/local/lib64/opensips/ and some in /usr/local/src/apps/opensips-1.6.1-notls/scripts/. Is there a suggested way to resolve this different so the database build script will work? Thanks ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips - using database views
Hi Oleg, most of the "id" columns are not used by opensips (neither written, nor read) - but they are used as primary keys. So, you can remove the columns, but be sure in setting another primary key for the tables. Regards, Bogdan Oleg Burlacu wrote: > Hi! > I'm planning to migrate from fixed database tables to views. > But I have a touble to generate the auto increment field - "id". > Can opensips operate with tables without the "id" filed? > > Thanks, > Oleg > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] high-availability - senario
I still have a problem with binding all IP some headers do not have the IP set : Record-Route: . Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bKfc58.1427b764.0. Is there a way to tell Opensip to use the routing source IP, or other alternative ? From: users-boun...@lists.opensips.org on behalf of Julien Chavanton Sent: Tue 02/02/2010 1:44 PM To: OpenSIPS users mailling list; OpenSIPS users mailling list; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] high-availability - senario This was well documented in opensips.cfg listen=udp:0.0.0.0:5060 or for both TCP/UDP listen=0.0.0.0:5060 From: users-boun...@lists.opensips.org on behalf of Julien Chavanton Sent: Tue 02/02/2010 12:47 PM To: OpenSIPS users mailling list; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] high-availability - senario I deceided not to use a loadbalancer because, the IP address for outbound call would have been different and we want it to be completely transparent for existing Interco. I went with heartbeat + mon I created sip.monitor When the fail over take effect it is not listening on the virtual IP, how do I configure opensips to bind to 0.0.0.0:5060 ? From: users-boun...@lists.opensips.org on behalf of Nigel Daniels Sent: Tue 26/01/2010 11:45 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] high-availability - senario have you considerd using keepalived with vrrp instead ? On Tue, Jan 26, 2010 at 7:03 AM, Julien Chavanton wrote: I am configuring high-availability with heart-beat + LVS + ldirector I do not want to load balance but mostly make sure it will fail over quickly, I have found situation where the in LVS (UDP/TCP) connection never timeout when the remote IP send periodic OPTIONS request for example. I beleive I will have to set very low UDP/TCP time-out, however with such a low time-out I can not load-balance so I will use weighted Round-Robin with very high priority 65535 on the active one and weith 1 on the passive server. Any other suggested way to cluster without load-balancing ? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Nigel Daniels Network & Systems Administrator ConnectAndSell inc. (650)-533-2542 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips - using database views
Hi Bogdan, Let's suppose that we have 2 tables, one for domains and other for users but with the domain field the foreign key from domains table. Also I have in the users table the bit type fields "local", "mobile"... In this case our output from the created view "grp" will be username domaingrp last_modified xxyy10sip1.abcd.efginternational1900-01-01 00:00:01 xxyy10sip1.abcd.efglocal1900-01-01 00:00:01 xxyy10sip1.abcd.efgmobile1900-01-01 00:00:01 xxyy10sip1.abcd.efgnational1900-01-01 00:00:01 So, each table has a primary key, but the "result" - not. Can I use this result for group module? Or, I should generate also a primary key? Best regards, Oleg On Tue, Feb 2, 2010 at 4:13 PM, Bogdan-Andrei Iancu wrote: > Hi Oleg, > > most of the "id" columns are not used by opensips (neither written, nor > read) - but they are used as primary keys. > > So, you can remove the columns, but be sure in setting another primary > key for the tables. > > Regards, > Bogdan > > Oleg Burlacu wrote: > > Hi! > > I'm planning to migrate from fixed database tables to views. > > But I have a touble to generate the auto increment field - "id". > > Can opensips operate with tables without the "id" filed? > > > > Thanks, > > Oleg > > > > > > ___ > > Users mailing list > > Users@lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > Bogdan-Andrei Iancu > www.voice-system.ro > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] high-availability - senario
On 02.02.2010 14:17, Julien Chavanton wrote: > I still have a problem with binding all IP some headers do not have the IP > set : > > Record-Route: . > Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bKfc58.1427b764.0. > > Is there a way to tell Opensip to use the routing source IP, or other > alternative ? There are two options, (never used the first one though - it might work): 1) advertised_address="1.2.3.4" (your real ip) 2) Set ip_nonlocal_bind via sysctl, which will allow you to bind to an interface even if it's not available on that host. You can have a hot standby host this way (on the virtual address), because it will start getting messages as soon as you update the routing. Just bind to the address you want instead of 0.0.0.0. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] priority differences between carrierroute and drouting
Seems that a weighted load balancing is not directly posible with drouting (http://lists.opensips.org/pipermail/users/2009-September/008106.html), but you can trick the system to behave like this. It would be great if the module could also support this feature natively, maybe by handling a special format on the gwlist colum (GW1=0.4,GW2=0.2,GW3=0.4) I was interested on use drouting, because it can be managed with opensips control panel. Is the carrierroute module on plans? 2010/2/1 Gabriel Bermudez > > Hi everybody, > > I've read the documentation on carrierroute and drouting, about the algorithm > used to select a gateway. I understand that on carrierroute the gateway > selected depends on the prefix dialed and the value of the prob field. For > example for prefix 49, gateway x.x.x.x with prob=0.2 and gateway y.y.y.y with > prob=0.8 can exists in the carrierroute table. > > ++-+---+-+---++---+ > | id | carrier | domain | scan_prefix | flags | prob | rewrite_host | > ++-+---+-+---++---+ > | 1 | 1 | 0 | 49 | 0 | 0.2 | x.x.x.x > | > | 2 | 1 | 0 | 49 | 0 | 0.8 | > y.y.y.y | > ++-+---+-+---++---+ > > I understand that this means 20% of the calls will be sent to x.x.x.x and 80% > to y.y.y.y > Is this the same for drouting, or only the highest priority rule is selected? > > Thanks for your answers > > Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] dns name and media_relay_avp.
Hi there. Is it posible to use dns-name at media_relay_avp? Now I just get error: media-dispatcher[356]: warning: user requested media_relay relay-01.test.local is not available -- WBR, Leonid Nasedkin cell: +7 923 731 0077 icq: 256816 skype: leonidnasedkin ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] lb_reload issue
Hello all, Is it possible to execute lb_reload command from remote machine using some java/c/c# program. Or any other way to achieve this. Pls help me out. -- Thanks Rajib Deka Software Engineer Servion Global Solution Chennai, India Mobile No: + 91 80157 09130 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] lb_reload issue
El Martes, 2 de Febrero de 2010, rajib deka escribió: > Hello all, > > Is it possible to execute lb_reload command from remote machine using some > java/c/c# program. Or any other way to achieve this. Pls help me out. Sure, use the Manager Interface remotely by using the mi_xmlrpc or mi_datagram modules. -- Iñaki Baz Castillo ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] RTP listener
Hello all, I would to listen and record RTP streams in real-time. RTP proxy seems to be able to record RTP streams in pcap format. Is there a way to listen RTP streams in real-time? Thanks for any help, Yannick ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Bulding OpenSIPS v1.6.1-notls on 64 bit machine
On 02.02.2010 16:10, Steven C. Blair wrote: > I'm trying to build a fresh copy of OpenSIPS v1.6.1-notls on a 64 bit RH > system and experiencing some path issues when running opensipsdbctl create. > It seems some scripts exist in /usr/local/lib64/opensips/ and some in > /usr/local/src/apps/opensips-1.6.1-notls/scripts/. Is there a suggested way > to resolve this different so the database build script will work? Steven, You should provide the exact console output that shows what happens. Personally I and many people here are using 1.6.1 on 64 bit machines. -- Sincerely, Andrew Pogrebennyk ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] What is protocol/port mismatch?
Hello list,
The problem was that the voicemail server was respecting the host
and port of the routeset which OpenSIPS was rewriting to acommodate
the transport change from TLS to TCP. The voicemail server was
configured to send over TLS to the outbound proxy however, so
traffic was arriving at OpenSIPS on the port 5060 using TLS.
It's clear now why the warnings were appearing in the logs, but
even more it seems that OpenSIPS sometimes crashes when sending
TLS traffic to a TCP port which it is expecting clear text from.
To know about the solution to this problem, read below.
An ven., janv 29, 2010, opensipsl...@encambio.com schrieb:
>An ven., janv 29, 2010, Bogdan-Andrei Iancu schrieb:
>>opensipsl...@encambio.com wrote:
>>> ...and I see this in the log:
>>>
>>>opensips[2717]: WARNING:core:get_send_socket: protocol/port
>>> mismatch
>>>
>>> some hundreds of times per day (about once every 20 minutes
>>> per registered UAC.)
>>>
>>> I have this in the route script:
>>>
>>> listen = tls:name.host.tld:5061
>>> [...]
>>> t_relay("name.host.tld:5080");
>>>
First, although all UAs were exchanging SIP traffic with OpenSIPS
over TLS, t_relay was relaying the messages over UDP. I assume that
t_relay is either hardcoded to use UDP or OpenSIPS was routing the
messages to itself internally over UDP (to recursively resolve
aliases.) In this case maybe t_relay implicitly uses the transport
over which the message was last received (even internally.)
In any case, the call to t_relay should look like this if relaying
over the TCP transport is desired:
t_relay("tcp:name.host.tld:5080");
The problem then is that the rr module will add two Record-Route
headers, expecting a symetrical fashion of SIP traffic. I turned
this off with modparam("rr", "enable_double_rr", 0) because the
voicemail server (Asterisk) is very tricky (maybe even buggy) to
configure to use TLS and outbound proxies.
At this point I saw that traffic was arriving at OpenSIPS port 5061
over TLS as expected, and as indicated by the routeset. The problem
is solved.
>Is the basic idea of this warning message that OpenSIPS
>exchanges a SIP message with a UA over a certain transport
>(TLS in this case) and port number (5061 in this case), but
>a t_relay in the route script forwards the message over a
>different transport or to a different port number?
>
The basic idea of this warning message is that OpenSIPS expects
a certain type (TLS or not TLS) of traffic at a certain TCP port
as configured by the 'listen' directives. If a UAC sends TLS
traffic to a port only configured to listen to plain text traffic
then this warning will appear. What happens to the mismatched
traffic is not clear, but I assume that it is ignored by OpenSIPS.
>If so, what is being compared and how is it compared?
>
The TCP port on which OpenSIPS listens to and the type of traffic,
either TLS encrypted or plain text.
Regards,
Brian
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Re: [OpenSIPS-Users] number of opensips children
Hello Bogdan, An mer., déc 23, 2009, Bogdan-Andrei Iancu schrieb: >opens...@encambio.com wrote: >> An ven., déc 18, 2009, Bogdan-Andrei Iancu schrieb: >>> To see what processes you have and what they are doing, do: >>> >>> opensipsctl fifo ps >>> >> # opensipsctl fifo ps >> Process:: ID=0 PID=24975 Type=attendant >> Process:: ID=1 PID=24977 Type=SIP receiver udp:123.234.210.1:5060 >> Process:: ID=2 PID=24978 Type=time_keeper >> Process:: ID=3 PID=24979 Type=timer >> Process:: ID=4 PID=24980 Type=MI FIFO >> >> [...] >> >> My gut feeling is that having four UDP listening processes and four >> TCP listening processes is about right for us, because we only have >> a handful of UACs participating infrequently (5 calls per day.) >> >Actually that is more than needed - during some performance tests (only >simply call relaying) we managed to put 6K cps in a single process. > I have eight TCP listeners configured and about sixteen UACs are connected. I get a ton of these warnings whenever REGISTER or INVITE messages come in: Feb 02 18:17:22 name.host.tld opensips[02126]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (1) Feb 02 18:17:25 name.host.tld opensips[02126]: WARNING:core:send2child: no free tcp receiver, connection passed to the leastbusy one (1) Because you mentioned that you benchmarked 6K CPS with a single process (was it TCP?), I'd like to know if you got as many warnings as well. One question is: What does 'free tcp receiver' mean? I assumed that listening TCP ports were free to accept as many connections as needed. By the way, each of the 16 UACs registered to the 8 TCP listener processes is avoiding NAT problems by keeping the TCP connection open by setting the tcp_persistent_flag. Is OpenSIPS expecting there to be at least one TCP listener process which is not encumbered by the tcp_persistent_flag? Regards, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Quota system problem
Hi all, I installed Opensips 1.6 + CDRTool version 6.9 and I'm not able to setup quota for postpaid users.I added the *quota* column in opensips.subscriber table. And after what else to setup? What is the change on opensips.cfg ? Is there any complete howto (tutorial) for quota/postpaid system? Can anyone help me?? -- Peers ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] number of opensips children
Hello,
An mar., févr 02, 2010, opensipsl...@encambio.com schrieb:
>An mer., déc 23, 2009, Bogdan-Andrei Iancu schrieb:
>>opens...@encambio.com wrote:
>>> An ven., déc 18, 2009, Bogdan-Andrei Iancu schrieb:
>>> My gut feeling is that having four UDP listening processes and four
>>> TCP listening processes is about right for us, because we only have
>>> a handful of UACs participating infrequently (5 calls per day.)
>>>
>>Actually that is more than needed - during some performance tests (only
>>simply call relaying) we managed to put 6K cps in a single process.
>>
>I have eight TCP listeners configured and about sixteen UACs are
>connected. I get a ton of these warnings whenever REGISTER or INVITE
>messages come in:
>
> Feb 02 18:17:22 name.host.tld opensips[02126]:
> WARNING:core:send2child: no free tcp receiver, connection passed to the
> leastbusy one (1)
> Feb 02 18:17:25 name.host.tld opensips[02126]:
> WARNING:core:send2child: no free tcp receiver, connection passed to the
> leastbusy one (1)
>
>Because you mentioned that you benchmarked 6K CPS with a single
>process (was it TCP?), I'd like to know if you got as many warnings
>as well. One question is:
>
> What does 'free tcp receiver' mean? I assumed that listening
> TCP ports were free to accept as many connections as needed.
>
> [...]
>
>Is OpenSIPS expecting there to be at least one TCP listener process
>which is not encumbered by the tcp_persistent_flag?
>
At risk of answering my own question and questioning my own answer,
I'd like to suggest the following change:
--- tcp_main.c.orig 2010-01-18 12:33:49.151095000 +0100
+++ tcp_main.c 2010-02-02 20:07:15.263065567 +0100
@@ -911,7 +911,7 @@
tcp_children[idx].busy++;
tcp_children[idx].n_reqs++;
if (min_busy){
- LM_WARN("no free tcp receiver, connection passed to the least"
+ LM_INFO("no free tcp receiver, connection passed to the least "
"busy one (%d)\n", min_busy);
}
LM_DBG("to tcp child %d %d(%d), %p\n", idx, tcp_children[idx].proc_no,
That would correct the defective english spelling 'leastbusy' as
well as ridding the log of a properly running OpenSIPS server of
false warnings. I'm assuming of course, that it's perfectly okay
for TCP listener processes to keep a TCP connection open by using
the tcp_persistent_flag and accept new SIP requests at the same
time.
Regards,
Brian
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Re: [OpenSIPS-Users] lb_reload issue
Thank you very much for your reply. I think this is what I was searching for. Let you know once I successfully apply it. Cheers Rajib On Tue, Feb 2, 2010 at 10:00 PM, Iñaki Baz Castillo wrote: > El Martes, 2 de Febrero de 2010, rajib deka escribió: > > Hello all, > > > > Is it possible to execute lb_reload command from remote machine using > some > > java/c/c# program. Or any other way to achieve this. Pls help me out. > > Sure, use the Manager Interface remotely by using the mi_xmlrpc or > mi_datagram > modules. > > -- > Iñaki Baz Castillo > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Rajib Deka Software Engineer Servion Global Solution Chennai, India Mobile No: + 91 80157 09130 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Quota system problem
Haven't worked with Quota yet, but perhaps this will get you started down the right path Look at the "bye_on_timeout_flag" modparam in Dialog module http://www.opensips.org/html/docs/modules/devel/dialog.html#bye-on-timeout-flag-id Heard about the "bye_on_timeout_flag" on this post http://lists.opensips.org/pipermail/users/2008-October/001116.html -- View this message in context: http://n2.nabble.com/Quota-system-problem-tp4502804p4503572.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Config Suggestions Request (SIP Trunks)
I've never once been able to get sipXecs to send out a register packet, the are very clear in the documentation that they expect a sip trunk. ie static configuration. Mike On 2/02/10 11:44 PM, Jeff Pyle wrote: > Mike, > > I have at least one customer on sipXecs. It registers just fine. Granted, > there is a bunch of other stuff messed up, but I think that's more my > customer's misunderstanding of how to configure it. > > > - Jeff > > > On Feb 2, 2010, at 1:51 AM, Mike O'Connor wrote: > > >> Hi All >> >> I've a have a couple of customers who are all asking to use sipXecs, >> which from my investigations does not support registrations. Instead it >> excepts that the ITSP provide unauthenticated trunks for inbound and >> outbound calls. >> >> So my question is what is the recommended way of supporting this ? >> >> Thanks >> Mike >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to force opensips add port field in via header?
Hi Bogdan and Iñaki, Thank you every much. Bogdan, Could you send me the patch? 2010/2/2 Iñaki Baz Castillo > El Martes, 2 de Febrero de 2010, Bogdan-Andrei Iancu escribió: > > Hi Iñaki, > > > > actually there is no param to force adding port to the local VIA when > > the port is the SIP default one (just checked the code). > > Sure? what about "advertised_port=5060"? :) > > Some years ago I used it in openser 1.2 to avoid a stupid bug in a stupid > gateway (Nortel CS2K, the worst machine in the world). > > > > -- > Iñaki Baz Castillo > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Lei.Tang lei.tl...@gmail.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Starting OpenXCAP without any logs
Havn't resloved the former problem,I got into the new trouble. The errors shows as follows,could you give me some suggestions? Is it the fault of python？？ r...@ubuntu:/var/log/openxcap# /usr/bin/openxcap --no-fork Starting OpenXCAP 1.1.2 xcap.server.HTTPFactory starting on 80 error: Traceback (most recent call last): error: File "/usr/lib/python2.5/threading.py", line 462, in __bootstrap error: self.__bootstrap_inner() error: File "/usr/lib/python2.5/threading.py", line 486, in __bootstrap_inner error: self.run() error: File "/usr/lib/python2.5/threading.py", line 446, in run error: self.__target(*self.__args, **self.__kwargs) error: --- --- error: File "/usr/lib/python2.5/site-packages/twisted/python/threadpool.py", line 210, in _worker error: result = context.call(ctx, function, *args, **kwargs) error: File "/usr/lib/python2.5/site-packages/twisted/python/context.py", line 59, in callWithContext error: return self.currentContext().callWithContext(ctx, func, *args, **kw) error: File "/usr/lib/python2.5/site-packages/twisted/python/context.py", line 37, in callWithContext error: return func(*args,**kw) error: File "/usr/lib/python2.5/site-packages/twisted/enterprise/adbapi.py", line 426, in _runInteraction error: conn = self.connectionFactory(self) error: File "/usr/lib/python2.5/site-packages/twisted/enterprise/adbapi.py", line 38, in __init__ error: self.reconnect() error: File "/usr/lib/python2.5/site-packages/twisted/enterprise/adbapi.py", line 75, in reconnect error: self._connection = self._pool.connect() error: File "/usr/lib/python2.5/site-packages/twisted/enterprise/adbapi.py", line 395, in connect error: conn = self.dbapi.connect(*self.connargs, **self.connkw) > > Yes. As I see you're running a hybrid system, because twisted version in > Lenny is 8.1.0 and you're running 9.0.0. This shouldn't matter, however > I need to test OpenXCAP with latest Debian unstable to check if changes > are needed because of updated library versions, I'll keep you posted. > > > Regards, > > -- > Saúl Ibarra Corretgé > AG Projects > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to force opensips add port field in via header?
Hi Bogdan, I tried to modify msg_translator.c, change line 1933 from if ((send_sock->port_no!=SIP_PORT) || (port_str!=&send_sock->port_no_str)) to if (1) It seem work now. 2010/2/3 Lei Tang > Hi Bogdan and Iñaki, Thank you every much. > Bogdan, Could you send me the patch? > > 2010/2/2 Iñaki Baz Castillo > > El Martes, 2 de Febrero de 2010, Bogdan-Andrei Iancu escribió: >> > Hi Iñaki, >> > >> > actually there is no param to force adding port to the local VIA when >> > the port is the SIP default one (just checked the code). >> >> Sure? what about "advertised_port=5060"? :) >> >> Some years ago I used it in openser 1.2 to avoid a stupid bug in a stupid >> gateway (Nortel CS2K, the worst machine in the world). >> >> >> >> -- >> Iñaki Baz Castillo >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > > -- > Lei.Tang > lei.tl...@gmail.com > -- Lei.Tang lei.tl...@gmail.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Config Suggestions Request (SIP Trunks)
Pardon my ignorance, but what exactly is the problem? Your customer does not need to register to you, however, then you need to have a static ip address to route to. and vice versa. When a customer sends an INVITE to your opensips, you need to authenticate by IP address. The sipXecs does not need to regsiter. Your customer will also have to enable ip authentication from you as well. Again, pardon my ignorance. Mike O'Connor wrote: > I've never once been able to get sipXecs to send out a register packet, > the are very clear in the documentation that they expect a sip trunk. ie > static configuration. > > Mike > > > > On 2/02/10 11:44 PM, Jeff Pyle wrote: > >> Mike, >> >> I have at least one customer on sipXecs. It registers just fine. Granted, >> there is a bunch of other stuff messed up, but I think that's more my >> customer's misunderstanding of how to configure it. >> >> >> - Jeff >> >> >> On Feb 2, 2010, at 1:51 AM, Mike O'Connor wrote: >> >> >> >>> Hi All >>> >>> I've a have a couple of customers who are all asking to use sipXecs, >>> which from my investigations does not support registrations. Instead it >>> excepts that the ITSP provide unauthenticated trunks for inbound and >>> outbound calls. >>> >>> So my question is what is the recommended way of supporting this ? >>> >>> Thanks >>> Mike >>> >>> ___ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] MediaProxy as bridge between Private and Public Interfaces
Hi All, I have MediaProxy working in a multi-homed setup where it is acting as a relay between an interface on a public IP and one on a private IP (We connect to our SIP provider over a private network via the private interface using static routes in the route table, and a SNAT out the private interface) The system is functioning and calls are being relayed between public and private interfaces, however with every call, I am receiving the following in the logs (which is a bit unnerving): Feb 3 15:19:32 SIPProxy1 media-relay[6806]: Traceback (most recent call last): Feb 3 15:19:32 SIPProxy1 media-relay[6806]: File "/usr/local/lib/python2.5/site-packages/twisted/internet/udp.py", line 126, in doRead Feb 3 15:19:32 SIPProxy1 media-relay[6806]: self.protocol.datagramReceived(data, addr) Feb 3 15:19:32 SIPProxy1 media-relay[6806]: File "/usr/local/lib/python2.5/site-packages/mediaproxy/mediacontrol.py", line 127, in datagramReceived Feb 3 15:19:32 SIPProxy1 media-relay[6806]: self.cb_func(host, port, data) Feb 3 15:19:32 SIPProxy1 media-relay[6806]: File "/usr/local/lib/python2.5/site-packages/mediaproxy/mediacontrol.py", line 203, in got_data Feb 3 15:19:32 SIPProxy1 media-relay[6806]: self.substream.check_create_conntrack() Feb 3 15:19:32 SIPProxy1 media-relay[6806]: File "/usr/local/lib/python2.5/site-packages/mediaproxy/mediacontrol.py", line 253, in check_create_conntrack Feb 3 15:19:32 SIPProxy1 media-relay[6806]: self.forwarding_rule = _conntrack.ForwardingRule(self.caller.remote, self.caller.local, self.callee.remote, self.callee.local, self.stream.session.mark) Feb 3 15:19:32 SIPProxy1 media-relay[6806]: Error: No such file or directory I have tried strace to determine what file may be missing or lacking permissions, however there are references to hundreds of files and it is proving difficult to track down. Has anyone seen this before, or could suggest some other way of troubleshooting the issue? Many thanks, Daniel Worrad Ipera Communications ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Config Suggestions Request (SIP Trunks)
Authenticate by IP Address, yep but how ? What is the recommend way of handling this and how do I route DID's to them. That was my question :) Mike On 3/02/10 3:50 PM, David J. wrote: > Pardon my ignorance, but what exactly is the problem? > > Your customer does not need to register to you, however, then you need > to have a static ip address to route to. > and vice versa. When a customer sends an INVITE to your opensips, you > need to authenticate by IP address. > > The sipXecs does not need to regsiter. > > Your customer will also have to enable ip authentication from you as well. > > Again, pardon my ignorance. > > > > Mike O'Connor wrote: > >> I've never once been able to get sipXecs to send out a register packet, >> the are very clear in the documentation that they expect a sip trunk. ie >> static configuration. >> >> Mike >> >> >> >> On 2/02/10 11:44 PM, Jeff Pyle wrote: >> >> >>> Mike, >>> >>> I have at least one customer on sipXecs. It registers just fine. Granted, >>> there is a bunch of other stuff messed up, but I think that's more my >>> customer's misunderstanding of how to configure it. >>> >>> >>> - Jeff >>> >>> >>> On Feb 2, 2010, at 1:51 AM, Mike O'Connor wrote: >>> >>> >>> >>> Hi All I've a have a couple of customers who are all asking to use sipXecs, which from my investigations does not support registrations. Instead it excepts that the ITSP provide unauthenticated trunks for inbound and outbound calls. So my question is what is the recommended way of supporting this ? Thanks Mike ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> ___ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
