sr-users November 2005

sr-users@lists.kamailio.org

245 participants
571 discussions

[Users] table version error
by Raymond Chen 23 Nov '05

23 Nov '05

Dear all, Start the CVS HEAD in debug mode gave me this error, and idea? 0(0) table_version(): No row for table re_grp found 0(0) ERROR:group:mod_init: Invalid table version for re_grp (use ser_mysql.sh reinstall) 0(0) init_mod(): Error while initializing module group ERROR: error while initializing modules 0(0) XLOG: destroy module ... 0(0) pool_remove: Removing connection from the pool 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done Ray

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[Users] RE: RE : Harry's list???
by harry gaillac 23 Nov '05

23 Nov '05

I don't swear my english is too poor . However some people are quite aggressive on these lists. Harry --- Olivier Taylor <olivier.taylor(a)gmail.com> a écrit : > And if they don't as quickly as you expect, you > swear :( > Not a good idea. > > Olivier > > -----Message d'origine----- > De : harry gaillac [mailto:gaillacharry@yahoo.fr] > Envoyé : mercredi 23 novembre 2005 13:20 > À : Olivier Taylor > Cc : serusers(a)iptel.org; users(a)openser.org > Objet : RE: Harry's list??? > > > Olivier, > > I hate "Sarkozy mode" However i guess some > experienced > people on these lists can tell me yes it's possible > or > not ! > > That's all I ask to the lists > > Regards Harry > > --- Olivier Taylor <olivier.taylor(a)gmail.com> a > écrit > : > > > There is something buggy in this list, where is > the > > bug reporting tool? > > > > Maybe it's time to stop that flooding now. > > > > Please, Harry, post your problems one by one and > > please be patient, you must > > know that the 'Sarkozy mode' is not really > > appreciated and will ot help. > > > > Kind regards, > > > > Olivier > > > > > > > > > > > > ___________________________________________________________________________ > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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Re: [Users] Re: Cesc ask for expelling Harry
by Yair Hakak 23 Nov '05

23 Nov '05

i quote from your email: Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 LOOKS LIKE THE SAME DAMN PORT TO ME. i tried to help you, i offered to help, but evidently you're more interested in being an ass than getting an answer to your question. here's one more person who might have helped you who you've pissed off and now won't even bother. -yair p.s. dont bother responding, you're on my blacklist. On 11/23/05, harry gaillac <gaillacharry(a)yahoo.fr> wrote: > > > > what about this is not clear? > > > > NO ONE CAN HELP YOU > > > > maybe it's because your problem is esoteric > > maybe it's because you're not explaining it well > > maybe it's because you continue to attempt to use > > idioms in english > > when it's clear that you have no understanding of > > them > > maybe it's because you frequently come across as > > threatening (as in > > "when i get an answer i'll stop posting") > > maybe it's because you cross-post > > > > it's up to you to make clear what you are looking > > for, and then to ASK > > FOR (as opposed to demanding) help. > > You're kidding I spend time to explain my problem > > > i, for one, use SER and asterisk together, SER as > > the SIP proxy and > > Asterisk as the dialplan server and PSTN gateway. I > > get SER to do the > > NAT transversal using mediaproxy and nathelper, and > > use the dialplan > > and gateway features in asterisk because of > > increased call control, > > cleaner dialplan logic, call logging, all that good > > stuff. If i > > understand you correctly, you want to use asterisk > > as a proxy (why, i > > don't know, it's not a great SIP proxy). If i can > > help, i will, but i > > have to first understand what you want to do. > > > > Thanks but I've ever done > > > for example, you claim to be running both asterisk > > and SER on the same > > box on the same port. unless i'm missing something > > with the NAT/public > > IP setup, that seems like a not such-a-good-idea. > > Not the same port > > > and stop sending diagrams. they only make it worse. > > > and, here is the golden rule, and you yourself say > > this: if you send a > > question and dont get an answer, DO NOT SEND THE > > SAME QUESTION AGAIN, > > because, surprisingly enough, you won't get an > > answer and you will > > piss people off. > > Here is an other golden rule > when you can help do it > > Regards > Harry > > > > > > > > ___________________________________________________________________________ > Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger > Téléchargez cette version sur http://fr.messenger.yahoo.com >

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Re: [Users] open letter (2)
by Klaus Darilion 23 Nov '05

23 Nov '05

harry gaillac wrote: >>Have you ever used SIP clients with presence and IM? >>I suggest to setup >>ser (without Asterisk) just to test the IM features. >>SIP based >>IM/presence implementations are very poor yet. > > I've done it And what were your experiences? Which clients do you use? >>In your picture, the NAT router is on the same PC as >>ser and asterisk. >>Is this correct? > > this is correct It would be a good idea to split things. This is a rather complicated setup. >>what scenario do you have? Are all the users behding >>the same NAT (in >>the same subnet) and you provide VoIP within this >>network (e.g. an >>enterprise) or do you have external users (e.g. like >>iptel or >>freeworlddialup)? > > in fact both > > > asterisk+ser > private net=====nathelper ======nat===private net > nat box > || > internet====== I suggest: 1. Asterisk, ser and the RTP proxy 8rtpproxy or mediaproxy) should listen only on the public interface (this really must be a routable public IP address, no private). 2. Setup the firewall (e.g. iptables) correctly to allow traffic from/to ser, asterisk and the RTP proxy 3. setup ser according the "getting started" document on onsip.org. AFAIK this document contains hints how to route to a gateway. Reuse this part of the config to route certain calls to the asterisk box. 4. Try to solve things step by step: - REGISTER should work fine from Internet and LAN - Calls from Internet clients to Internet clients - Calls from LAN clients to LAN clients - Calls from LAN clients to Internet clients (and vice versa) - now try to add asterisk, e.g. calling a certain number will be routed to asterisk and starts the echo application If all the above works (DO NOT start integrating the asterisk as long as basic SIP call do not work!!!!!), you can implement your setup. 5. Do really read every word in the "getting started" document, if things are unclear read it again. 6. Do not post "how to make this setup". Ask small questions addressing particular (small) problems. 7. Post to the related list. - do not post to developer lists - if you use ser, post to ser's list - if you use openser, post to openser's list - if you have an asterisk problem, ask at the asterisk list (e.g. you want to solve NAT traversal and registration with ser. Thus, do not ask this kind of questions at the asterisk list). 8. always remember that this support is voluntary 9. If you don't find the proper english word, look into the dictionary instead of using another word which might also have other meanings. 10. Go and buy an english SIP book. (this will you help to learn the english terms for all the SIP stuff) 11. use ngrep to watch the SIP call flow # ngrep -t -d any port 5060 regards klaus

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[Serusers] Custom headers in SER
by Andrey Kouprianov 23 Nov '05

23 Nov '05

Hello: I am using SER 0.9.3. Whenever I attach a custom header to REGISTER request, the response would come back without it. It looks like SER strips all unknown headers from the request, before generating response. Is there any way to keep custom headers in my responses? Andrey.

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[Serusers] Network topology
by Kyriakos Mavromichalis 23 Nov '05

23 Nov '05

Hi, Im trying to figure out what the network topology should be for my SIP network. What I want to do is basically to divide my users into 3 subnets and assign a SER proxy with a public ip for each subnet. The domain should be the same for all users ie sip,otenet-telecom.com and every time a user would send a register message the dns resolution would point him to the proxy that I chose to serve the users of the subnet the user to be registered belongs to. Can this be done or will I need to define separate domains ie sip1.otenet-telecom.com for the first subnet sip2.otenet-telecom.com for the second subnet etc... -- Kyriakos Mavromichalis Otenet Telecom

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[Serusers] Removing SUBSCRIBE info from SER
by Jimmy Montano 23 Nov '05

23 Nov '05

Greetings I've been trying to remove SUBSCRIBE info from SIP to stop softphone users to see one another using peer-to-peer presence Anyone knows how can this be accomplished? Thanks in Advance

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[Users] redirect - timer C
by Andréa Giordanna 22 Nov '05

22 Nov '05

Hi all, How can I configure openser to send a 302 message only when timer C fires? Is it possible? Thanks, Giordanna --------------------------------- Yahoo! Acesso Grátis: Internet rápida e grátis. Instale o discador agora!

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Re: [Users] open letter
by harry gaillac 22 Nov '05

22 Nov '05

In fact ser should keep nat opened of ua behind nat. Ser just need to keep location for im an presence Asterisk forward requests according to contact field to ser. --- Iqbal <iqbal(a)gigo.co.uk> a écrit : > Okay, so get ser to fix the NAT part before sending > to asterisk. Any is > ser just proxying all register commands, why not > register in ser, than > asterisk, I know you are doing IM in asterisk, and I > havent done that, Asterisk do not support IM/presence. > but I am using asterisk for features like call > pickup and transfer, they > might be different in operation but I think its best > to find out howto > let ser do all the hardwork and let asterisk only > work when it needs to. They can work together ! thanks for help harry > harry gaillac wrote: > > >not exactly ! > > > >something like this : > > > > asterisk > > | > > ser > > ua1| | ua2 > > > > > >ua1 and ua2 send registration to asterisk via ser . > > > >when ua1 invite ua2 sip INVITE is sent to ser > which > >one forward it to asterisk. > >asterisk lookup in its AORs so it bridge the call > and > >send INVITE to ua2 via ser. > > > >Harry > >--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > > > > > > > >>Okay almost there :-) > >> > >>So UA ---> asterisk ---> SER ---> UA > >> > >>is that it > >> > >>harry gaillac wrote: > >> > >> > >> > >>> > >>> > >>> > >>> > >>>>okay, so ALL your users are registering to > >>>>asterisk...is that correct. > >>>> > >>>> > >>>> > >>>> > >>>Correct via ser as outbound sip proxy > >>> > >>> > >>> > >>> > >>>>If so the problem is howto accept users from > >>>> > >>>> > >>behind > >> > >> > >>>>a NAT into asterisk, > >>>>or am I confusing things further. > >>>> > >>>> > >>>> > >>>> > >>>the problem is in contact field. > >>>when user agents send register we have in sip hf > >>>Contact sip:user@privateip > >>>So asterisk store this AOR and try to contact > agent > >>>via nat box instead of SER > >>> > >>> > >>> > >>> > >>> > >>>>If the above are true, where is SER in this, or > >>>> > >>>> > >>are > >> > >> > >>>>users hitting SER > >>>>and you are sending the REGISTER from ser into > >>>>asterisk. > >>>> > >>>> > >>>> > >>>> > >>>SER is an outbound sip proxy which handle IM > >>> > >>> > >>presence > >> > >> > >>>nat > >>> > >>>Harry > >>> > >>> > >>> > >>> > >>> > >>>>>>> One box > >>>>>>> --------------------------- > >>>>>>> | ---------------- | > >>>>>>> | | asterisk pbx | | > >>>>>>> | ---------------- | > >>>>>>> | || | > >>>>>>> | ---------- ---------- > >>>>>>> | | SER |====|NAT box |==== private > >>>>>>> | ---------- ---------- > >>>>>>> |-------------------------- > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>___________________________________________________________________________ > >> > >> > >> > >>>Appel audio GRATUIT partout dans le monde avec le > >>> > >>> > >>nouveau Yahoo! Messenger > >> > >> > >>>Téléchargez cette version sur > >>> > >>> > >>http://fr.messenger.yahoo.com > >> > >> > >>>. > >>> > >>> > >>> > >>> > >>> > > > > > > > > > > > > > > > >___________________________________________________________________________ > > >Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > >Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > > >. > > > > > > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Users] Inbound failover from third-party PSTN
by Daryl Sanders 22 Nov '05

22 Nov '05

Greeting Everyone, I hit a bit of a road block in my design. I'm using SRV for proxy failover on the customer side, but I'm wondering if third-party PSTN providers will support SRV when delivering calls to my SERs. I want to make sure if the PSTN provider cannot reach one SER they try another. I've only talked to a couple providers so far, and it seems they want to just send traffic to an IP address. Is this the norm? Is HA with heartbeat and a single IP the only solution here? Thanks! - Daryl

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sr-users November 2005