Re: [asterisk-users] extension with callerid not found in context
Thank you, Joshua! -- sergio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extension with callerid not found in context
I have pjsip endpoint with callerid= context=localpeers which
looks follow:
[localpeers]
exten => _.@_./sergio,1,Dial(Local/${EXTEN}@somecontext)
exten =>_.@_.,1,NoOp()
And this works fine:
== Setting global variable 'SIPDOMAIN' to 'DOMAIN'
-- Executing [EXTEN@localpeers:1]
Dial("PJSIP/pjsip_endpoint-000a", "Local/EXTEN@somecontext") in new
stack
But this:
[localpeers]
exten => _.@_./sergio,1,Dial(Local/${EXTEN}@somecontext)
fails:
res_pjsip_session.c:2993 new_invite: Call from 'pjsip_endpoint'
(TLS:IP:PORT) to extension 'EXTEN' rejected because extension not found
in context 'localpeers'.
Why do I need `exten => _.@_.,1,NoOp()` record?
--
sergio.
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Re: [asterisk-users] linphone calls not missed due to cause not 487
On 16/10/2020 10:11, Michael Maier wrote: Sometimes, linphone shows missed calls as missed. You could try to reproduce it I can't reproduce it, it happens less than once a month. -- sergio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] linphone calls not missed due to cause not 487
Sometimes, linphone shows missed calls as missed. Look like asterisk replies with cause=487 that time, but I can't understand why. Grandstream always shows calls as missed ones. How should I investigate this? -- sergio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] linphone calls not missed due to cause not 487
Hello. Calls cancelled by caller during the dialing phase, are shown in Linphone as simply past calls, not missed ones. I thought this is an Linphone issue, but Sylvain says it's on my PBX side: https://github.com/BelledonneCommunications/linphone-android/issues/832#issuecomment-557474864 > The CANCEL message has a Reason header with Q.850 protocol and cause 0, which doesn't mean call has been missed (should be 487). Is this my dialplan / setup or an Asterisk issue? How can I get Asterisk to send cause=487? -- sergio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] some domains resolving issues
We could extend that to runtime as well. Would be nice! -- sergio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] some domains resolving issues
It was due to a lack of tcp or udp sections with transport declaration in pjsip.conf. But it's still unclear, 1. How should I find this? Is a log so poor and needs to be reported, or am I missing something? 2. Why I need to set bind? I use this transport only for outgoing connections. -- sergio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] some domains resolving issues
On 30/09/2020 14:59, Joshua C. Colp wrote: latest version of 16 on Ubuntu 16.12.0~dfsg-1 ? -- sergio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] some domains resolving issues
on target '_sips._tcp.sip.linphone.org' res_pjsip/pjsip_resolver.c:177 sip_resolve_add: [0x7f4e740593f8] Added target 'sip6.linphone.org' with record type '1', transport 'TLS', and port '443' res_pjsip/pjsip_resolver.c:349 sip_resolve_callback: [0x7f4e740593f8] SRV record received on target '_sip._tcp.sip.linphone.org' res_pjsip/pjsip_resolver.c:349 sip_resolve_callback: [0x7f4e740593f8] SRV record received on target '_sip._tcp.sip.linphone.org' res_pjsip/pjsip_resolver.c:349 sip_resolve_callback: [0x7f4e740593f8] SRV record received on target '_sip._udp.sip.linphone.org' res_pjsip/pjsip_resolver.c:349 sip_resolve_callback: [0x7f4e740593f8] SRV record received on target '_sip._udp.sip.linphone.org' res_pjsip/pjsip_resolver.c:413 sip_resolve_callback: [0x7f4e740593f8] New queries added, performing parallel resolution again res_pjsip/pjsip_resolver.c:277 sip_resolve_callback: [0x7f4e740593f8] All parallel queries completed res_pjsip/pjsip_resolver.c:326 sip_resolve_callback: [0x7f4e740593f8] A record received on target 'sip6.linphone.org' res_pjsip/pjsip_resolver.c:326 sip_resolve_callback: [0x7f4e740593f8] A record received on target 'sip1.linphone.org' res_pjsip/pjsip_resolver.c:326 sip_resolve_callback: [0x7f4e740593f8] A record received on target 'sip6.linphone.org' res_pjsip/pjsip_resolver.c:419 sip_resolve_callback: [0x7f4e740593f8] Resolution completed - 3 viable targets res_pjsip/pjsip_resolver.c:201 sip_resolve_invoke_user_callback: [0x7f4e740593f8] Address '0' is 54.37.202.229:5223 with transport 'TLS' res_pjsip/pjsip_resolver.c:201 sip_resolve_invoke_user_callback: [0x7f4e740593f8] Address '1' is 91.121.209.194:5223 with transport 'TLS' res_pjsip/pjsip_resolver.c:201 sip_resolve_invoke_user_callback: [0x7f4e740593f8] Address '2' is 54.37.202.229:443 with transport 'TLS' res_pjsip/pjsip_resolver.c:207 sip_resolve_invoke_user_callback: [0x7f4e740593f8] Invoking user callback with '3' addresses % host iptel.org iptel.org has address 212.79.111.155 iptel.org mail is handled by 50 mx3.zoho.com. iptel.org mail is handled by 10 mx.zoho.com. iptel.org mail is handled by 20 mx2.zoho.com. % host -t SRV _sip._tcp.iptel.org _sip._tcp.iptel.org has SRV record 0 100 5060 sip.iptel.org. % host -t SRV _sip._udp.iptel.org _sip._udp.iptel.org has SRV record 0 25 5060 sip.iptel.org. % host sip.iptel.org sip.iptel.org has address 212.79.111.155 I've already tried to ask community.asterisk.org without success. https://community.asterisk.org/t/resolving-issue/85861 -- sergio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] domain name in TO header
Hello. I have a problem with linphone: https://github.com/BelledonneCommunications/linphone-android/issues/583 It creates different dialogues for different TO headers. And asterisk uses client IP in TO header, so I have a new dialogue each time my phone changes IP address. There is a from_domain option that is specified in my pjsip.conf, but I've not found any analogue for to header. But I've found same (unanswered) question on community.asterisk.org: https://community.asterisk.org/t/asterisk-pjsip-how-to-force-using-domain-name-instead-of-ip-address-for-to-header/71898 -- sergio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WSS ISSUE
Hello, Open the url in the browser "https://xxx:8089/ws " address, you accept the ssl certificate and retry the sip registry. With that in principle ought to be solved Best regards Sergio Virviescas. Telf: +34 722557601 Email: developersavs...@gmail.com / svirvies...@novationits.com De: asterisk-users-boun...@lists.digium.com <asterisk-users-boun...@lists.digium.com> en nombre de Антон Сацкий <satski...@gmail.com> Enviado: lunes, 23 de mayo de 2016 14:55:17 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] WSS ISSUE HI all can anybody help me there to search a problem from time to time "Connection closed before receiving a handshake response" WebSocket connection to 'wss://XXX:8089/ws' failed: Connection closed before receiving a handshake response sipml.js?14636613801642821:16763 ws_close (index):2731 failed_to_start - Failed to connet to the server sipml.js?14636613801642821:16764 WebSocket connection to 'wss://X:8089/ws' failed: Connection closed before receiving a handshake response -- Best regards Antony tel. +380669197533 tel2. +380636564340 Paypal http://paypal.me/Satskiy<http://paypal.me/Satskiy?ppid=PPC000654=PL=en_PL(en_DK)=NN8XJS9XEP22C=21db79ac-ef8d-11e5-9553-9c8e992ea258==4d776c21ca7d2=4d776c21ca7d2=4d776c21ca7d2_tpcid=ppme-social-business-profile-created=main:email=main:email=op=em=ci=sys> satski...@gmail.com<mailto:mail%3asatski...@gmail.com> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] seems like call is picked and returned to me
Hi all I hope that someone of you can solve this. Right now I'm stuck! I'm using asterisk with some SIP extensions. Basically I want to establish a call between desktop voip phone (ext 181) and embedded sip system (ext 182) All I can see in CLI is: == Using SIP RTP CoS mark 5 -- Executing [182@default:1] Dial(SIP/181-000a, SIP/182) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/182 -- SIP/182-000b is ringing -- SIP/182-000b is making progress passing it to SIP/181-000a -- SIP/182-000b answered SIP/181-000a -- Remotely bridging SIP/181-000a and SIP/182-000b == Spawn extension (default, 182, 1) exited non-zero on 'SIP/181-000a' Seems like extension 182 (called ext) is getting call and passing them another time to me 181 (origin call) I've try it with siemens pbx and works as expected cheers! Sergio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8, busylevel and CCBS
My question is so complex and I try to explain well.
We have a customer that he wants limits incoming calls to his extensions
to only one. That's not complicated with GROUPCOUNT, DEVICE_STATE or
SIPPEER with curcalls option.But the problem is when you want implement
CCBS service.
If we have next context:
exten=_XXX,1,NOOP()
same=n,GotoIF($[${DEVICE_STATE(${ARG2})}=BUSY]?occupied)
same=n,Dial(SIP/${EXTEN})
same=n,GotoIf($[${DIALSTATUS}=BUSY]?ocupado)
same=n,Hangup()
same=n(occupied),Busy()
same=n,Hangup()
If we call to 100 extensions and that extensions reject call or no
answer call, we can use CallCompletionRequets to request CCNR service
and all work fine.
But when a call is on 100 extension, and you call to 100 extension and
go to occupied label, if you reques a CCBS with
CallCompletionRequest() this application fails with NO_CORE_INSTANCE
error.
It's appear like CCSS only work with DIALSTATUS variable and with Dial
application I don't know how to limit to only one incoming call.
Are there any way to solve this?
Any help would be appreciated.
regards,
Sergio
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Re: [asterisk-users] Postgresql in Asterisk
I finally solve the problem, in gentoo the permission of dir /var/run/postgresql/ is: drwxrwx--- 2 postgres postgres 4096 Feb 29 18:09 postgresql so if we want to connect asterisk to postgresql, we need to add the user that runs asterisk to the group postgres and with this finally I can connect with unixODBC to postgresql database I hope this help some one. Regards, On Mon, 2012-02-27 at 13:49 -0600, Sergio Basurto wrote: Thank you Jonathan, I already do the steps you mention, my configuration is: in res_odbc.conf enabled = yes dsn = asterisk-connector pre-connect = yes in odbc.ini [asterisk-connector] Description = PostgreSQL connection to 'asterisk' database Driver = PostgreSQL Database= db_asterisk Servername= localhost UserName= asterisk Password= secret Port= 5432 Protocol= 9.1 ReadOnly= No RowVersioning = No ShowSystemTables= No ShowOidColumn = No FakeOidIndex= No ConnSettings= in odbcinst.ini [PostgreSQL] Description = ODBC for PostgreSQL Driver = /usr/lib/libodbcpsql.so Setup = /usr/lib/libodbcpsql.so FileUsage = 1 if I run with root: #echo select 1 | isql -v asterisk-connector returns +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ SQL select 1 ++ | ?column? | ++ | 1 | ++ SQLRowCount returns 1 1 rows fetched This show me that it can connect, the thing is that in the asterisk logs it returns: res_odbc.c: Connecting asterisk res_odbc.c: res_odbc: Error SQLConnect=-1 errno=101 [unixODBC]Could not connect to the server; Could not connect to remote socket res_odbc.c: Failed to connect to asterisk res_odbc.c: Registered ODBC class 'asterisk' dsn-[asterisk-connector] res_odbc.c: res_odbc loaded. I notice that if I run the isql command with other user than root, it returns [S1000][unixODBC]Could not connect to the server; Could not connect to remote socket. [ISQL]ERROR: Could not SQLConnect I guess is an extra configuration for ODBC that I am missing, what you think? Regards, On Fri, 2012-02-24 at 13:16 -0600, Jonathan Rose wrote: You need to make sure ODBC is actually getting a connection made with your database. What you should see under ODBC DSN settings: Name: asterisk DSN:asterisk-connector Last connection attempt: WHATEVER Pooled: No/Yes Connected: Yes Connected: Yes is the important part. Remember, you need to have an account in postgres that can be logged into. I made one on my machine with the following: name = asterisk password = secret And in /etc/odbc.ini, I have the following connector established: [asterisk-connector] Description = PostgreSQL connection to 'asterisk' database Driver = PostgreSQL Database= asterisk Servername = localhost UserName= asterisk Password= secret Port= 5432 Protocol= 8.1 I'm guessing this will be 9.1 in your case ReadOnly= No RowVersioning = No ShowSystemTables= No ShowOidColumn = No FakeOidIndex= No ConnSettings= While my res_odbc.conf looks like this: [asterisk] enabled = yes dsn = asterisk pre-connect = yes In addition to having a connector defined, you need to have an ODBC adapter for postgres. I think this might come with ODBC byd efault though. When I was using mysql, I had to get a separate adapter to make it work and set the path to it in Driver. I don't think that is the case with pgsql though. Go ahead and post your extconfig.conf. I'm guessing that the reason you are able to post CDRs in spite of not having the Connected status show up in your ODBC show is because you are connecting with res_pgsql.conf instead of odbc. - Original Message - From: Sergio Basurto sbasu...@soft-gator.com To: asterisk-users@lists.digium.com Sent: Wednesday, February 22, 2012 6:54:47 AM Subject: Re: [asterisk-users] Postgresql in Asterisk On Wed, 2012-02-22 at 06:48 -0600, Sergio Basurto wrote: Hello, I install asterisk an postgresql 9.1 in gentoo, I already did the configuration in both asterisk and postgresql, in fact If I make a call and asterisk log it to CDR table, my question is: I make a typo mistake I mean If I make a call asterisk already log it into CDR table
Re: [asterisk-users] Postgresql in Asterisk
Thank you Jonathan, I already do the steps you mention, my configuration is: in res_odbc.conf enabled = yes dsn = asterisk-connector pre-connect = yes in odbc.ini [asterisk-connector] Description = PostgreSQL connection to 'asterisk' database Driver = PostgreSQL Database= db_asterisk Servername= localhost UserName= asterisk Password= secret Port= 5432 Protocol= 9.1 ReadOnly= No RowVersioning = No ShowSystemTables= No ShowOidColumn = No FakeOidIndex= No ConnSettings= in odbcinst.ini [PostgreSQL] Description = ODBC for PostgreSQL Driver = /usr/lib/libodbcpsql.so Setup = /usr/lib/libodbcpsql.so FileUsage = 1 if I run with root: #echo select 1 | isql -v asterisk-connector returns +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ SQL select 1 ++ | ?column? | ++ | 1 | ++ SQLRowCount returns 1 1 rows fetched This show me that it can connect, the thing is that in the asterisk logs it returns: res_odbc.c: Connecting asterisk res_odbc.c: res_odbc: Error SQLConnect=-1 errno=101 [unixODBC]Could not connect to the server; Could not connect to remote socket res_odbc.c: Failed to connect to asterisk res_odbc.c: Registered ODBC class 'asterisk' dsn-[asterisk-connector] res_odbc.c: res_odbc loaded. I notice that if I run the isql command with other user than root, it returns [S1000][unixODBC]Could not connect to the server; Could not connect to remote socket. [ISQL]ERROR: Could not SQLConnect I guess is an extra configuration for ODBC that I am missing, what you think? Regards, On Fri, 2012-02-24 at 13:16 -0600, Jonathan Rose wrote: You need to make sure ODBC is actually getting a connection made with your database. What you should see under ODBC DSN settings: Name: asterisk DSN:asterisk-connector Last connection attempt: WHATEVER Pooled: No/Yes Connected: Yes Connected: Yes is the important part. Remember, you need to have an account in postgres that can be logged into. I made one on my machine with the following: name = asterisk password = secret And in /etc/odbc.ini, I have the following connector established: [asterisk-connector] Description = PostgreSQL connection to 'asterisk' database Driver = PostgreSQL Database= asterisk Servername = localhost UserName= asterisk Password= secret Port= 5432 Protocol= 8.1 I'm guessing this will be 9.1 in your case ReadOnly= No RowVersioning = No ShowSystemTables= No ShowOidColumn = No FakeOidIndex= No ConnSettings= While my res_odbc.conf looks like this: [asterisk] enabled = yes dsn = asterisk pre-connect = yes In addition to having a connector defined, you need to have an ODBC adapter for postgres. I think this might come with ODBC byd efault though. When I was using mysql, I had to get a separate adapter to make it work and set the path to it in Driver. I don't think that is the case with pgsql though. Go ahead and post your extconfig.conf. I'm guessing that the reason you are able to post CDRs in spite of not having the Connected status show up in your ODBC show is because you are connecting with res_pgsql.conf instead of odbc. - Original Message - From: Sergio Basurto sbasu...@soft-gator.com To: asterisk-users@lists.digium.com Sent: Wednesday, February 22, 2012 6:54:47 AM Subject: Re: [asterisk-users] Postgresql in Asterisk On Wed, 2012-02-22 at 06:48 -0600, Sergio Basurto wrote: Hello, I install asterisk an postgresql 9.1 in gentoo, I already did the configuration in both asterisk and postgresql, in fact If I make a call and asterisk log it to CDR table, my question is: I make a typo mistake I mean If I make a call asterisk already log it into CDR table. how can I make a function like the ones in func_odbc.conf for postgresql, if I am using res_pgsql.conf instead of res_odbc.conf? I also configure odbc and it connects with echo select 1 | isql -v asterisk-connector with out problems, but when I try an odbc function or restart asterisk it logs: Error SQLConnect=-1 errno=101 [unixODBC]Could not connect to the server; Could not connect to remote socket. and the command CLI odbc show ODBC DSN Settings - Name: asterisk DSN: asterisk-connector Last connection attempt: 2012-02-22 06
[asterisk-users] Postgresql in Asterisk
Hello, I install asterisk an postgresql 9.1 in gentoo, I already did the configuration in both asterisk and postgresql, in fact If I make a call and asterisk log it to CDR table, my question is: how can I make a function like the ones in func_odbc.conf for postgresql, if I am using res_pgsql.conf instead of res_odbc.conf? I also configure odbc and it connects with echo select 1 | isql -v asterisk-connector with out problems, but when I try an odbc function or restart asterisk it logs: Error SQLConnect=-1 errno=101 [unixODBC]Could not connect to the server; Could not connect to remote socket. and the command CLI odbc show ODBC DSN Settings - Name: asterisk DSN:asterisk-connector Last connection attempt: 2012-02-22 06:45:36 I will appreciate any help. Regards, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Postgresql in Asterisk
On Wed, 2012-02-22 at 06:48 -0600, Sergio Basurto wrote: Hello, I install asterisk an postgresql 9.1 in gentoo, I already did the configuration in both asterisk and postgresql, in fact If I make a call and asterisk log it to CDR table, my question is: I make a typo mistake I mean If I make a call asterisk already log it into CDR table. how can I make a function like the ones in func_odbc.conf for postgresql, if I am using res_pgsql.conf instead of res_odbc.conf? I also configure odbc and it connects with echo select 1 | isql -v asterisk-connector with out problems, but when I try an odbc function or restart asterisk it logs: Error SQLConnect=-1 errno=101 [unixODBC]Could not connect to the server; Could not connect to remote socket. and the command CLI odbc show ODBC DSN Settings - Name: asterisk DSN:asterisk-connector Last connection attempt: 2012-02-22 06:45:36 I will appreciate any help. Regards, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sergio Basurto sbasu...@soft-gator.com Soft Gator S.A de C.V. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to get Call-ID SIP header outside chan_sip scope ...
Hello there! I'm working on some modifications on Asterisk to adapt it to our needs considering some particular demandings of the infraestructure we want to provide. Two of these modifications are: 1- A proprietary configuration driver that will communicate with a server that will be the source of information for the entire infraestructure; and, 2- A call control application that will be responsible for call timing control and pre-paid support; Here we are prioritizing internal modifications and loadable modules (like modules, applications, etc) against external AGI components to acchieve the best performance possible for the entire solution. One problem we have here is to find out the best option (even one that results on some internal Asterisk files changing) that allow us to propagate the SIP header Call-ID to both modules described above. The best shot we have until now is to use the callid field from the sip_pvt structure of SIP channel, what will lead us to two considerable code changes: 1- Propagate the channel to method realtime_var_get of our proprietary ARA driver; and 2- Duplication of necessary structs to a header (.h) file so the modules can navigate on private structure sip_pvt. The first change isn't big deal. But the need of validation of the second modification, every time we make a merge with updated codes is concerning me a lot. Does anyone have a better approach to get this done ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get Call-ID SIP header outside chan_sip scope ...
Hello there! I really hate when this happens, but... It seems channel variable SIPCALLID will have the info I need, so the changes will be reduced to propagate the channel to ARA driver method realtime_var_get. If someone have any additional info, or can indicate some problem on using this variable, please let me know. Thanks and best regards, Mauro. Mauro Sergio Ferreira Brasil escreveu: Hello there! I'm working on some modifications on Asterisk to adapt it to our needs considering some particular demandings of the infraestructure we want to provide. Two of these modifications are: 1- A proprietary configuration driver that will communicate with a server that will be the source of information for the entire infraestructure; and, 2- A call control application that will be responsible for call timing control and pre-paid support; Here we are prioritizing internal modifications and loadable modules (like modules, applications, etc) against external AGI components to acchieve the best performance possible for the entire solution. One problem we have here is to find out the best option (even one that results on some internal Asterisk files changing) that allow us to propagate the SIP header Call-ID to both modules described above. The best shot we have until now is to use the callid field from the sip_pvt structure of SIP channel, what will lead us to two considerable code changes: 1- Propagate the channel to method realtime_var_get of our proprietary ARA driver; and 2- Duplication of necessary structs to a header (.h) file so the modules can navigate on private structure sip_pvt. The first change isn't big deal. But the need of validation of the second modification, every time we make a merge with updated codes is concerning me a lot. Does anyone have a better approach to get this done ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does L(x:y:z) Dial option work on Asterisk version 1.4 ?
Sorry guys.
My bad!
As you can see, the command on prior message is incorret.
I've changed to:
Dial(SIP/${EXTEN}|20|RtTL(30:6:2))
and it's working now.
Thanks and best regards,
Mauro.
Mauro Sergio Ferreira Brasil escreveu:
Hello there!
I'm testing Dial call limit option on Asterisk version 1.4.26, but
it's not working.
The issued command is: Dial(SIP/${EXTEN}|20|RtT|L(30:6:2)).
Am I missing something ?
Does it only work with Asterisk version 1.6.X ?
Thanks and best regards,
--
__At.,
_
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br mailto:@tqi.com.br
: www.tqi.com.br http://www.tqi.com.br
( + 55 (34)3291-1700
( + 55 (34)9971-2572
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[asterisk-users] [UOL - Manutenões Desktop] Controlling call duration ...
Hello there! The only available way to control call duration is using the RTCC patch (discussed here https://issues.asterisk.org/view.php?id=6335; and mainteined here http://ast.varna.net/;) ? The purpouse is to have a way to monitor (probably on a per-minute basis) and hangup costly calls (and/or multiple calls initiated by same SIP user). Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does L(x:y:z) Dial option work on Asterisk version 1.4 ?
Hello there!
I'm testing Dial call limit option on Asterisk version 1.4.26, but
it's not working.
The issued command is: Dial(SIP/${EXTEN}|20|RtT|L(30:6:2)).
Am I missing something ?
Does it only work with Asterisk version 1.6.X ?
Thanks and best regards,
--
__At.,
_
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br mailto:@tqi.com.br
: www.tqi.com.br http://www.tqi.com.br
( + 55 (34)3291-1700
( + 55 (34)9971-2572
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Re: [asterisk-users] Multiple user registration ...
Thanks a lot Faheem for you help.
I totaly understand now the approach you've used.
It's very interesting and inventive for sure.
I didn't know that I could append IP:Port info on user when using the
Dial command and that this will make calling to two different devices
registered using same user work.
With this little but extemelly important peace of information you gave
me the answer to our questions here.
Thanks again, and best regards,
Mauro.
Faheem escreveu:
The purpose of Perl script is to store user registrations records only
and nothing else regarding call dialing.
The script will main records like this.
User1:
IP1: 192.168.0.100 Por1: 5060
IP2: 69.30.21.10 Port2: 5060
User2:
IP1: 192.168.10.1 Por1: 5060
IP2: 192.168.10.1 Por2: 5061
User3:
IP1: 192.168.10.121 Por1: 5060
IP2: 192.168.10.123 Por2: 5061
and so on
No it all depends on you to store these information on files or database.
Assume you have stored IP/Ports in the database.
Database=cloneline
Table = users(username,ip1,port1,ip2,port2)
For dialing:
Assume username=user1 and extension =123456
exten= 123456,1,NoOp()
exten= 123456,n,MYSQL(Connect connid 'localhost' cdr dbpass cloneline)
exten= 123456,n,NoOP(Connection ID:${connid})
exten= 123456,n,MYSQL(Query resultid ${connid} SELECT\ ip1\, port1\,
ip2\, port2\, status\ from\ users\ where\ username=user1 )
exten= 123456,n,MYSQL(Fetch fetchid ${resultid} ip1 port1 ip2 port2)
exten= 123456,n,Dial(SIP/us...@${ip1}:${port1}SIP/us...@${ip2}:${port2})
for dialing user3
username=user3 and extension =112233
exten= 112233,1,NoOp()
exten= 112233,n,MYSQL(Connect connid 'localhost' cdr dbpass cloneline)
exten= 112233,n,NoOP(Connection ID:${connid})
exten= 112233,n,MYSQL(Query resultid ${connid} SELECT\ ip1\, port1\,
ip2\, port2\, status\ from\ users\ where\ username=user3 )
exten= 112233,n,MYSQL(Fetch fetchid ${resultid} ip1 port1 ip2 port2)
exten= 112233,n,Dial(SIP/us...@${ip1}:${port1}SIP/us...@${ip2}:${port2})
Hope every thing would be clear...
Muhammad Faheem
Software Engineer
AxVoice Inc.
307,Y Commercial,
DHA Lahore, Pakistan
+92-333-4793314
http://www.axvoice.com
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--
__At.,
_
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br mailto:@tqi.com.br
: www.tqi.com.br http://www.tqi.com.br
( + 55 (34)3291-1700
( + 55 (34)9971-2572
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Re: [asterisk-users] Multiple user registration ...
Thank you very much for all your help, Muhammad! (please let me know if I should call you Faheem, instead). I'll make some tests with this script on my premises as soon as possible. Having a look on it, I couldn't realize how it really works in conjunction with Asterisk. I mean, it seems that the line cloning is acchieved by the creation/update of a file (with a name that matches the SIP user name) inside folder /var/lib/asterisk/users. The point is that I couldn't find any similar folder on my test server, and a search on Google by this folder didn't returned any usefull results. Am I missing something here ? Suppose I want to acchieve this feature by database update. I've noticed here that it will be a problem considering that field name at sip_buddies, that is my Realtime table for SIP users, have a UNIQUE_KEY constraint. Moreover, I don't know what will happen on Realtime (probably an error or undesired behavior) that seems to be expecting just one record user record information. Have you tried database approach ? Thanks again and best regards, Mauro. Faheem escreveu: Mauro, Yes, you will receive simultaneous ring on all devices which are registered with the same SIP User Account. If a SIP user is registered on multiple devices i.e. only one SIP account is used and only one extension is used here in my implementation, then he will ring on all registered SIP enabled devices/softphones. Also I've tested it with following combinations of SIP enabled devices/Softphones. 1) Both ports of SPA2100 are registered with one SIP account(Same IP address but different ports) 2) The same SIP user is registered with one port of SAP2100 and the same user is registered with Xten (multiple IP addresses) 3) The same SIP User is registered with two different SIP Dialers. Here in these three cases I've sucessfully able to receive concurrent ring on the registered devices/softphones. Also CDR are working correctly. The perl script works perfectly with my customization, you need to modify it according to your requirements. Muhammad Faheem Software Engineer AxVoice Inc. 307,Y Commercial, DHA Lahore, Pakistan +92-333-4793314 http://www.axvoice.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple user registration ...
Outch... my bad. I saw Muhammad Faheem at the end of your email, and... well... thought the first was your name... sorry about that. Thank's a lot again, but I'm still curious about how Asterisk integrates with your secondary persistence. I mean... until now I saw only the codes regarding the persistence or multiple registration info, but I can't still realize how Asterisk perform the invitation to all these devices... Could you please explain how it works with your solution ? Thanks and best regards, Mauro. Faheem escreveu: Yes, Its my Name! Well, my DB server and asterisk servers are on different locations. For optimization I've used Files instead of Database queries. Secondly the /var/lib/asterisk/user folder is a simple folder if it does not exists on your asterisk machine then simple create it on the specified location or simply change the folder path in the perl script. Before File handling I've used Databases for maintaing active registered users with multiple IP/Ports. The attatched perl script uses database for maintain active registration. The structure of cloneline table should be. DB: Cloneline table:users(Username,IP1,Port1,Ip2,Port2) all varchars(30) Please adjust the table fields appropriately. Hope this code block will solve you problems. Muhammad Faheem Software Engineer AxVoice Inc. 307,Y Commercial, DHA Lahore, Pakistan +92-333-4793314 http://www.axvoice.com --- On *Fri, 8/28/09, Mauro Sergio Ferreira Brasil /mauro.bra...@tqi.com.br/* wrote: From: Mauro Sergio Ferreira Brasil mauro.bra...@tqi.com.br Subject: Re: [asterisk-users] Multiple user registration ... To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, August 28, 2009, 5:38 PM Thank you very much for all your help, Muhammad! (please let me know if I should call you Faheem, instead). I'll make some tests with this script on my premises as soon as possible. Having a look on it, I couldn't realize how it really works in conjunction with Asterisk. I mean, it seems that the line cloning is acchieved by the creation/update of a file (with a name that matches the SIP user name) inside folder /var/lib/asterisk/users. The point is that I couldn't find any similar folder on my test server, and a search on Google by this folder didn't returned any usefull results. Am I missing something here ? Suppose I want to acchieve this feature by database update. I've noticed here that it will be a problem considering that field name at sip_buddies, that is my Realtime table for SIP users, have a UNIQUE_KEY constraint. Moreover, I don't know what will happen on Realtime (probably an error or undesired behavior) that seems to be expecting just one record user record information. Have you tried database approach ? Thanks again and best regards, Mauro. Faheem escreveu: Mauro, Yes, you will receive simultaneous ring on all devices which are registered with the same SIP User Account. If a SIP user is registered on multiple devices i.e. only one SIP account is used and only one extension is used here in my implementation, then he will ring on all registered SIP enabled devices/softphones. Also I've tested it with following combinations of SIP enabled devices/Softphones. 1) Both ports of SPA2100 are registered with one SIP account(Same IP address but different ports) 2) The same SIP user is registered with one port of SAP2100 and the same user is registered with Xten (multiple IP addresses) 3) The same SIP User is registered with two different SIP Dialers. Here in these three cases I've sucessfully able to receive concurrent ring on the registered devices/softphones. Also CDR are working correctly. The perl script works perfectly with my customization, you need to modify it according to your requirements. Muhammad Faheem Software Engineer AxVoice Inc. 307,Y Commercial, DHA Lahore, Pakistan +92-333-4793314 http://www.axvoice.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At
Re: [asterisk-users] Multiple user registration ...
Hi Muhammad, and thanks a lot for the answer. On this moment I'm making some tests in order to collect enough information to participate of a meeting at the end of this day regarding the use of Asterisk. I won't have time to validate your contribution before this meeting and this info would be very handfull. So... could you please just clarify me if this approach you've used allows multiple SIP clients (softphone, ATA, VoIP-Celular) registrate with Asterisk using the same SIP user (like SIP/101, for example) on such way that if someone call this number all clients gets simultaneously called? Thanks and best regards, Mauro. Faheem escreveu: Dear Mauro, Your requirement seems Clone line feature for asterisk. The same question I've asked here in this group, a months later but could't get well. But actually implemented it now! It is done using AMI. Here is its basic psudo code. # ami-event.pl Connect to AMI Read the AMI Events Parse the events If it is registration Event then store the Username/IP/Ports/Technology in Database # dial plan run agi script to get all strings eg. first Device: SIP/u...@192.168.0.123:5061 second Device: SIP/u...@10.0.0.150:6060 The complete script is attached. Muhammad Faheem Software Engineer AxVoice Inc. 307,Y Commercial, DHA Lahore, Pakistan +92-333-4793314 http://www.axvoice.com http://advcomm.net/ --- On *Wed, 8/26/09, Mauro Sergio Ferreira Brasil /mauro.bra...@tqi.com.br/* wrote: From: Mauro Sergio Ferreira Brasil mauro.bra...@tqi.com.br Subject: [asterisk-users] Multiple user registration ... To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, August 26, 2009, 7:07 PM Hello there! We are planning to use Asterisk on our VoIP platform, and we are spending some brains on a way to provide the following facility: let some SIP user (extension) registrate with more than one client (ATA, SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate calls from any of this devices that are registrated with the same user - no problems on tests too -, but also receive INVITE requests on all devices if someone calls this user - yeah... here the thing gets creepy. The demand is quite simple: let a user registrate with multiple devices using the same SIP user on such way that if someone call him, all these registered devices will ring and the first to take the call will be the lucky one. The demand, as I've said, is quite simple and logical (translated to our living world), but the reality is a very different history. On our tests, always is the last registered application/device that receives the call indication. And only the last one. We are making some tests trying to kind of deceive Asterisk on second, third, and additional, registrations so it receives from Realtime fake extensions numbers on such a way that we can use all these fake extensions to build a queue dinamicaly (through ARA) and provide the desired ring on all functionality. I think this will lead us to lots of SIP sinalization and multi user registration problems, but that was the best shot we had here until now. I would like to know if anyone had the same demand and, maybe, have found any viable solution to it. Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br /mc/compose?to=mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At., _ *Technology and Quality on Information* Mauro Sérgio
Re: [asterisk-users] Realtime with rtcachefriends=no problems...
Thanks Atis, its working pretty fine now. Best regards, Mauro. Atis Lezdins escreveu: On Wed, Aug 26, 2009 at 12:11 AM, Mauro Sergio Ferreira Brasilmauro.bra...@tqi.com.br wrote: Hello there! Problem found. For some reason, the update statement below is generated with an invalid atribution of empty value '' to field port that is an integer. Because of that, this record keeps with prior fullcontact information that was updated by another client (which uses a different port) what leads to wrong client rtp packets routing... wow... that was weird... :-) [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime: Query: UPDATE sip_buddies SET fullcontact = '', ipaddr = '', port = '', regseconds = '0', username = '', regserver = '' WHERE name = '101' [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect integer value: '' for column 'port' at row 1 First of all... my appologies by the false alarm. But now I need your help to identify why is this update statement being generated wrongly. Does someone have any idea ? Asterisk Realtime Architecutre currently treats all fields as strings. I wish too that it would take into account actual field type retrieved from DESCRIBE statement and add the quotes only if it's string. You can safely do ALTER TABLE sip_buddies CHANGE COLUMN port port VARCHAR(5); Regards, Atis -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple user registration ...
Hello there! We are planning to use Asterisk on our VoIP platform, and we are spending some brains on a way to provide the following facility: let some SIP user (extension) registrate with more than one client (ATA, SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate calls from any of this devices that are registrated with the same user - no problems on tests too -, but also receive INVITE requests on all devices if someone calls this user - yeah... here the thing gets creepy. The demand is quite simple: let a user registrate with multiple devices using the same SIP user on such way that if someone call him, all these registered devices will ring and the first to take the call will be the lucky one. The demand, as I've said, is quite simple and logical (translated to our living world), but the reality is a very different history. On our tests, always is the last registered application/device that receives the call indication. And only the last one. We are making some tests trying to kind of deceive Asterisk on second, third, and additional, registrations so it receives from Realtime fake extensions numbers on such a way that we can use all these fake extensions to build a queue dinamicaly (through ARA) and provide the desired ring on all functionality. I think this will lead us to lots of SIP sinalization and multi user registration problems, but that was the best shot we had here until now. I would like to know if anyone had the same demand and, maybe, have found any viable solution to it. Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple user registration ...
Hi Elliot, and thanks for the reply. I'm not completely sure you've considered that the SIP users registered on all devices are the same. Have you ? I mean... How will I use Dial command with a sequence of same devices, like: Dial(SIP/101SIP/101SIP/101), for example ? That's why we are testing the possibility to create virtual devices on subsequent registrations, so we can at the end make something like: Dial(SIP/101SIP/101-001SIP/101-002) if someone dials to SIP/101. Note: SIP/101-001 and SIP/101-002 don't really exist. They will be provided by our ARA driver to allow the multiple device ringing. Thanks and best regards, Mauro. Elliot Otchet escreveu: Is your goal here to have multiple devices ring when an extension is dialed and the first one to answer take the call? If so, see the Dial command Dial(Technology/resourceTechnology/resourceTechnology/resource...[|timeout][|options][|URL]). When multiple technology/resource entries are listed, the first one to answer will take the call. That accomplishes your goal, if I understand you correctly. The nice part about doing it this way (with each device independently registered) is that you gain a substantial amount of granularity in controlling where calls go and you don't have to find creative ways (read: unsupported) to trick Asterisk or endpoints. If you're developing your own GUI to have people set up their devices, you can easily create a wizard that walks them through setting up each device and associating them together through either channel variables or other tables in a database. I use this methodology in 1.4 and it works quite reliably. For a good reference, check out http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial or from your Asterisk console try: 'core show application dialenter' It's not perfect because you can have devices that do funny things with a SIP INVITE, but in most cases it works very well. Regards, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple user registration ...
Hi Barry, and thanks for the reply! This was the first question I've made on meeting yesterday to decide about this facility. Having me here today making this question should give you an idea of the level of acceptance of my suggestion :-). Anyway, the idea is really try to make it work with only one SIP user. I totally agree with you that this is an unnatural behavior, but I have to agree as well with our commercial staff because their vision was naturaly translated from our telephony world (we don't have a different ID - telephone number - to each phone we have home, right ?). So, I thank you for your handy Dial approach, which will be easier than the queue approach I was considering before. Given that I'll acchieve the virtual devices running. Considering my annoying insistence on work with just one SIP user, do you have any helpfull thoughts to share that can help me out ? Best regards, Mauro. Barry L. Kline escreveu: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Instead of trying to make Asterisk do this unnatural act, why not register each device with a separate id, then use the dial function to call all of them? e.g.exten = 122,1,Dial(SIP/1SIP/2SIP/3) You could use some creating scripting to decide which devices to ring based on the dialed extension. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKlU65CFu3bIiwtTARAu0DAJ4szfX1dp/BNZojIKhgIL/tIhkjvQCeLXCf A+Dys6+LgrNhL/zQpU8Vuwk= =1Y6q -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple user registration ...
Thanks again Elliot for everything. Considering our needs to develop a proprietary ARA driver, I think it's possible to use it and make Asterisk believe that additional registrations of same SIP device are in fact different device registrations. BUT, and yes it's a big BUT, we will end with an Asterisk version a litle hacked and even if we get this working on some version now, it doesn't give us any guarantee that it will in future. Anyway, I've put this question here just to be sure no one has already made such a thing before, and how odd is it. I'll take care and don't use - on sip device names... thanks for that too. Best regards, Mauro. Elliot Otchet escreveu: Your first example illustrates why having multiple devices registered as the same entity is a bad idea. It is impossible to differentiate between each device when you have multiple registering as the same entity. My users also really like setting up rules per device/per caller. When you treat a group of devices as one, you make it really hard to do that. On your theoretical virtual devices in Asterisk - you either have a device or you don't. The device will need to register in order to receive a call, so if you're expecting to do some magic on the registration to have a user who registers with the credentials of user 101 and be assigned to user 101-001, you'll be disappointed in the results. Also, you'll want to steer away from using hyphens in your sip device names. Hyphens are used in the SIP channel driver for a special purpose and using them in your device names may cause problems. See http://www.digium.com/handbook-draft.pdf page 19 for more info. If you're looking for a good separator, try using the underscore (_) character instead. All that being said, if you want to register multiple devices with a single set of credentials, you might want to check out a SIP Proxy instead of Asterisk's SIP B2BUA. Some can handle multiple registrations with a single set of credentials quite nicely. Regards, Elliot -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio Ferreira Brasil Sent: Wednesday, August 26, 2009 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple user registration ... Hi Elliot, and thanks for the reply. I'm not completely sure you've considered that the SIP users registered on all devices are the same. Have you ? I mean... How will I use Dial command with a sequence of same devices, like: Dial(SIP/101SIP/101SIP/101), for example ? That's why we are testing the possibility to create virtual devices on subsequent registrations, so we can at the end make something like: Dial(SIP/101SIP/101-001SIP/101-002) if someone dials to SIP/101. Note: SIP/101-001 and SIP/101-002 don't really exist. They will be provided by our ARA driver to allow the multiple device ringing. Thanks and best regards, Mauro. Elliot Otchet escreveu: Is your goal here to have multiple devices ring when an extension is dialed and the first one to answer take the call? If so, see the Dial command Dial(Technology/resourceTechnology/resourceTechnology/resource...[|timeout][|options][|URL]). When multiple technology/resource entries are listed, the first one to answer will take the call. That accomplishes your goal, if I understand you correctly. The nice part about doing it this way (with each device independently registered) is that you gain a substantial amount of granularity in controlling where calls go and you don't have to find creative ways (read: unsupported) to trick Asterisk or endpoints. If you're developing your own GUI to have people set up their devices, you can easily create a wizard that walks them through setting up each device and associating them together through either channel variables or other tables in a database. I use this methodology in 1.4 and it works quite reliably. For a good reference, check out http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial or from your Asterisk console try: 'core show application dialenter' It's not perfect because you can have devices that do funny things with a SIP INVITE, but in most cases it works very well. Regards, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971
Re: [asterisk-users] Multiple user registration ...
Thanks again Barry for the help and attention. Thanks for wishing me lucky as well... If we insist on this road I'll need it for sure :-). I can't agree more with your position, and I'll try to be sure our commercial demands can't be acchieved with normal approaches before adventuring on such path. Best regards, Mauro. Barry L. Kline escreveu: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Well, our phones at home are probably analog and can be connected in parallel. Unfortunately, VoIP phones are a different matter and need to be identified individually. I guess I don't get the problem your commercial side is having with this concept. You can produce the same result doing things within the constraints of SIP using the features built into Asterisk. Doing what you want may be possible with a bunch of contortions, but it's going to be an unnatural act fraught with tons of unexpected behavior. If you do get it working the way you describe you'll likely be doing so because of a side-effect behavior in a GIVEN version of Asterisk. The moment you change versions, the side effect may or may not be the same and you may find yourself in the same trouble. I can't offer anything more to help you except to wish you the best of luck. You're going to need it. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKlWonCFu3bIiwtTARAu1WAJ0eS2Eh6n6Tici9eDA82UIesuozNACaA9yi jT8u2aZfUHcSXGvJnc1FDEI= =VQhJ -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime with rtcachefriends=no problems...
Hello there! I was testing Asterisk for the last two weeks using the Realtime driver for MySQL, and leaving rtcachefriends=yes configured to enable MWI. Today I started making additional tests with rtcachefriends=no because we will probably need to use Asterisk without this cache. For some strange reason, calls stop to get routed between the SIP clients. I've registered successfuly with two sip clients as usual, but the call indication that I have on originator client (call in progress) don't match with the target client that indicates nothing at all. Using Wireshark I could see lots of ICMP errors being returned from the target machine with Destination Unreachable/Port Unreachable indications. And this happens on both ways, client 1 calling client 2 and vice-versa. I switched back to rtcachefriends=yes and all worked fine again. (note: always I change rtcachefriends to no, I change qualify parameter of all SIP users to no as well - to avoid warnings on CLI). Does anyone had this problem ? What Am I missing here ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime with rtcachefriends=no problems...
Hello there! Problem found. For some reason, the update statement below is generated with an invalid atribution of empty value '' to field port that is an integer. Because of that, this record keeps with prior fullcontact information that was updated by another client (which uses a different port) what leads to wrong client rtp packets routing... wow... that was weird... :-) [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime: Query: UPDATE sip_buddies SET fullcontact = '', ipaddr = '', port = '', regseconds = '0', username = '', regserver = '' WHERE name = '101' [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect integer value: '' for column 'port' at row 1 First of all... my appologies by the false alarm. But now I need your help to identify why is this update statement being generated wrongly. Does someone have any idea ? Thanks and best regards, Mauro. Mauro Sergio Ferreira Brasil escreveu: Hello there! I was testing Asterisk for the last two weeks using the Realtime driver for MySQL, and leaving rtcachefriends=yes configured to enable MWI. Today I started making additional tests with rtcachefriends=no because we will probably need to use Asterisk without this cache. For some strange reason, calls stop to get routed between the SIP clients. I've registered successfuly with two sip clients as usual, but the call indication that I have on originator client (call in progress) don't match with the target client that indicates nothing at all. Using Wireshark I could see lots of ICMP errors being returned from the target machine with Destination Unreachable/Port Unreachable indications. And this happens on both ways, client 1 calling client 2 and vice-versa. I switched back to rtcachefriends=yes and all worked fine again. (note: always I change rtcachefriends to no, I change qualify parameter of all SIP users to no as well - to avoid warnings on CLI). Does anyone had this problem ? What Am I missing here ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call routing between two Asterisk boxes using SIP not working ...
Hello there! I need some help to configure two Asterix boxes to route calls using SIP. I followed the instructions present at this site: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html;, but I couldn't get it working so far. The only difference, besides the names that I've used, is that I'm using realtime to retrieve all information. Both boxes registrate on the other perfectly. The problem happens when one call gets routed. It seems that realtime on destination box is trying to find locally a SIP user 1001 that is the originator of the call and is a user of the original box. It finally ends with a: chan_sip.c:14780 handle_request_invite: Failed to authenticate user 1001 sip:1...@10.10.100.158;tag=as1e79b629 on destination box. Wireshark present on destination box indicates all the following steps: 1- Wengo client registered with user 1001 starts the call to number 2001 with Box 1 (at 10.10.100.158); 2- Box 1 makes the challenge; 3- Wengo replies the challenge; 4- Box 1 send an successfull ack to Wengo client and sends the INVITE to Box 2 (at 10.10.100.156) that holds user 2001; 5- Box 2 makes the challenge; 6- Box 1 replies the challenge; 7- Box 2 sends a 403 Forbidden; Has anyone had this problem ? Can anyone help me out on that ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call routing between two Asterisk boxes using SIP not working ...
Hi guys! The problem was solved by the use of same password for registration users of both boxes. Is there no way to indicate different password for registration user of Box1 and registration user of Box2 ? Thanks and best regards, Mauro. Mauro Sergio Ferreira Brasil escreveu: Hello there! I need some help to configure two Asterix boxes to route calls using SIP. I followed the instructions present at this site: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html;, but I couldn't get it working so far. The only difference, besides the names that I've used, is that I'm using realtime to retrieve all information. Both boxes registrate on the other perfectly. The problem happens when one call gets routed. It seems that realtime on destination box is trying to find locally a SIP user 1001 that is the originator of the call and is a user of the original box. It finally ends with a: chan_sip.c:14780 handle_request_invite: Failed to authenticate user 1001 sip:1...@10.10.100.158;tag=as1e79b629 on destination box. Wireshark present on destination box indicates all the following steps: 1- Wengo client registered with user 1001 starts the call to number 2001 with Box 1 (at 10.10.100.158); 2- Box 1 makes the challenge; 3- Wengo replies the challenge; 4- Box 1 send an successfull ack to Wengo client and sends the INVITE to Box 2 (at 10.10.100.156) that holds user 2001; 5- Box 2 makes the challenge; 6- Box 1 replies the challenge; 7- Box 2 sends a 403 Forbidden; Has anyone had this problem ? Can anyone help me out on that ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Platform decision ...
Hello there! During some research on Internet I found the following comparison on site Voip-Info (see, http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ;): The main points listed on Asterisk's CONS that concerned me were: * Conferencing on Asterisk depends on Zaptel hardware and/or kernel modules for timing; * Lack of built-in STUN support for SIP NAT traversal; * Asterisk doesn't use SpanDSP; * Use of no longer maintained Berkeley DB1 engine as its internal database; * Asterisk doesn't allow CSRC entries in RTP; * Asterisk doesn't have an universal jitterbuffer for use with any channel type; * Asterisk doesn't use POSIX realtime extensions (having dependency with Zaptel timing); We were considering Asterisk as the chosen platform, but after reading this I got a little worried. The comparison considers 1.4 old version of Asterisk. So, can someone give me an update on what have changed for this items considering new 1.6 version ? Maybe someone can point me a site with an updated comparison. As long as I could see by now SpanDSP is present on new version of Asterisk, so this item isn't a difference any more. Right ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Platform decision ...
Man... I need to be very frank with you... I don't know any more. We started analysing what can be done to get Asterisk working on a way we want it to work, that is: totally dynamic dial plan generated by an external server (responsible for business logic and legacy interface), and retrieved through an new configuration driver (something like res_config_legacy.c). This point is clear to us now that is reachable without much effort. We considered, at first, a infraestructure with a redirect-server/load-balancer (played by OpenSIPS) directing the voip calls to final Asterisk instances. The problem is that after getting the first issue solved (about the driver acessing the legacy interface explained above), I started a research about Asterisk scalability and I didn't liked of what I found. Consulting some friends of mine that work with Voip (but that unfortunatelly don't need the PBX features) the impression was worst. One of them told me that on the only part of their infraestructure where Asterisk is used they want at all costs to remove it. Making things short, I need to have sure that Asterisk can handle a considerable number of concurrent calls, or I need an indication of another PBX that is scalable to be placed on Asterisk's place and that can be changed to retrieve the dialplan (or what it uses on call routing) from another server. Does anyone have any idea ? Thanks and best regards, Mauro. C. Savinovich escreveu: It all depends what are you going to use Asterisk for. Sounds like it is for conferencing. Would you care to elaborate? CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio Ferreira Brasil Sent: Tuesday, August 18, 2009 10:23 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Platform decision ... Hello there! During some research on Internet I found the following comparison on site Voip-Info (see, http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ;): The main points listed on Asterisk's CONS that concerned me were: * Conferencing on Asterisk depends on Zaptel hardware and/or kernel modules for timing; * Lack of built-in STUN support for SIP NAT traversal; * Asterisk doesn't use SpanDSP; * Use of no longer maintained Berkeley DB1 engine as its internal database; * Asterisk doesn't allow CSRC entries in RTP; * Asterisk doesn't have an universal jitterbuffer for use with any channel type; * Asterisk doesn't use POSIX realtime extensions (having dependency with Zaptel timing); We were considering Asterisk as the chosen platform, but after reading this I got a little worried. The comparison considers 1.4 old version of Asterisk. So, can someone give me an update on what have changed for this items considering new 1.6 version ? Maybe someone can point me a site with an updated comparison. As long as I could see by now SpanDSP is present on new version of Asterisk, so this item isn't a difference any more. Right ? Thanks and best regards, -- __At., _ *Technology and Quality on Information* Mauro Sérgio Ferreira Brasil Coordenador de Projetos e Analista de Sistemas + mauro.bra...@tqi.com.br mailto:@tqi.com.br : www.tqi.com.br http://www.tqi.com.br ( + 55 (34)3291-1700 ( + 55 (34)9971-2572 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
A similar issue happens to us. Make sure that, for inbound AND outbound calls rtp packets are reaching the other endpoint. If a NAT device(s) is between the endpoints make sure that the device NATs the traffic on BOTH ways (inbound AND outbound). Regards On Saturday 27 September 2008 23:54:37 Philip Prindeville wrote: I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way voice (I can hear the caller but they can't hear me). I've looked at tcpdumps of the RTP traffic, and the addresses and port numbers correspond to what's in the SIP INVITE/OK messages (assuming that they don't somehow get munged by NAT after tcpdump looks at them -- there is no NAT device upstream of my Asterisk firewall). I'll look into using Record() or Monitor() to capture the phone call, but if there's any conversion being done by codecs then that won't eliminate the possibility that the code itself is misconfigured or buggy and generating a bad stream on one of the legs... Anyone have an idea about how to best go about troubleshooting this? Thanks, -Philip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] nokia e51 (Christian Lox)
Hi Christian, I have been using the Nokia E51 with asterisk for a month now without any problems. It took me a while to configure it. I downloaded from Nokia a file (dont remember the name now, I am not on my pc at them moment) that added more features such as g729 etc. it is working great. My asterisk is on a public ip address, maybe that helps. Take care, -- Sergio Fabian Veltri Director Business IT Of: +54-11-5217-1297 Ext. 2201 Cell: +54-911-5977-0977 http://www.businessit.biz IT Service Management and Control Best Practices -- Message: 5 Date: Sun, 20 Jan 2008 00:10:58 +0100 From: Christian Lox [EMAIL PROTECTED] Subject: [asterisk-users] nokia e51 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-15; format=flowed Hi all. Anyone to share some experience with Nokia E51 and asterisk? We are trying to connect the E51 to our asterisk but to no avail. Googling said that it should work, but we are seeing real strange things here: - tcpdump reveals the nokia is talking to other ports than 5060 - registration is not possible at all, right now there is no network traffic to the asterisk box at all. A softphone on the same wlan segment registers without any problem. The how-tos on the web suggest different settings concerning the proxy/registration setupBut none of them works for us. But we are not nokia guys at all So, any help greatly appreciated! The setup: Cisco AP with EAP-TLS. Connected to an switch on which several vlans are connected to a cisco router. The internal network (192.168.23.0/24) talks to the DMZ, on which the radius (for EAP-TLS) and also the asterisk box is hosted. IP Addresses are assigned via DHCP from the AP. The Laptop from which i am writing has x-lite installed and that works just fine with the same credentials we are trying to setup the nokia: 2001 abc sipgate No RFC3581 We have been playing with nat=yes|no, but we cant get it to work. Thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP RealTime Friends
Hi, Do you know how see the peers statuses like: sip show peers but when sip peers are configured by Relatime method. Thanks 0xception escribió: yes you can use the type friend On 5/3/07, *Forrest Beck* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Con Netfono, puede hablar por telefono, de PC a PC y gratis ! Instale su Netfono desde http://www.netfono.com. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Interconnecting Cisco 1760 routers with Asterisk
Check this out HYPERLINK
javascript:ol('http://www.voip-info.org/wiki-Asterisk+cisco+FXO');http://w
ww.voip-info.org/wiki-Asterisk+cisco+FXO
_
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Joesph
Enviado el: Martes, 03 de Abril de 2007 02:53 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [asterisk-users] Interconnecting Cisco 1760 routers with Asterisk
Good day everyone.
I have Cisco 1760 routers that do site to site voip. Each router has 2 fxs
ports that connect to the local pbx and use sip to connect to other routers
over the WAN. I am thinking of putting in an asterisk box at the hub site
for interconnectivity with our global office voip provider. This provider
runs asterisk.
Question is - can Cisco 1760 routers make/receive calls to/fro asterisk? if
yes, any sample configuration please?
Thanks and regards
Joesph
Abuja, Nigeria
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RE: [asterisk-users] Re: Marks SNMP HowTo
How can i see if snmp is running ok on mi * box ? Thanks in advance -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Forrest Beck Enviado el: Domingo, 25 de Febrero de 2007 06:14 p.m. Para: Asterisk Users List Asunto: [asterisk-users] Re: Marks SNMP HowTo OK. problem solved. It was something dumb on my part. /var/agentx didn't have enough permissions to let asterisk access the socket. On 2/25/07, Forrest Beck [EMAIL PROTECTED] wrote: I followed Marks SNMP howto on Voip Magazine and ran into a small problem... (http://www.voip-magazine.com/content/view/2877/0/1/3/) When asterisk is running as a non-root user (asterisk) SNMP request for for the Asterisk MIB tree return nothing. If I quit asterisk and run it as root, all is fine. Does anyone have a idea what is going on? I have never used agentX, so I am unsure of what it is doing. Does it bind to a particular port that maybe my asterisk user does not have permission to access??? Here is my snmpd.conf file: master agentx agentXPerms 0660 0550 asterisk asterisk com2sec local localhost da_public com2sec mynetwork 10.11.0.0/16 da_public com2sec dmz 172.17.0.0/16 da_public group MyROGroup any local group MyROGroup any mynetwork group MyROGroup any dmz view all included .1 access MyROGroup any noauth 0 all none none and here is res_snmp.conf [general] subagent = yes enabled = yes Thanks all.! -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.441 / Virus Database: 268.18.3/699 - Release Date: 23/02/2007 01:26 p.m. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.4/702 - Release Date: 25/02/2007 03:16 p.m. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Didn't get a frame from channel
Using tdm400. While transfering a call from outside to another extensions, while this outside call is waiting with music, the another extension call hangs up suddenly, and the call is back to the outside call suddenly. Wathcing logs: Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462 bytes of audio while expecting 640 Jan 15 13:32:55 DEBUG[27850] channel.c: Didn't get a frame from channel: SIP/219-081d4d60 Jan 15 13:32:55 DEBUG[27850] channel.c: Bridge stops bridging channels SIP/219-081d4d60 and Zap/1-1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Hangup: channel: 1 index = 0, normal = 16, callwait = -1, thirdcall = -1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: disabled echo cancellation on channel 1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Updated conferencing on 1, with 0 conference users 15 13:32:55 VERBOSE[27850] logger.c: -- Hungup 'Zap/1-1' Jan 15 13:32:55 DEBUG[27850] app_dial.c: Exiting with DIALSTATUS=ANSWER. Jan 15 13:32:55 VERBOSE[27850] logger.c: == Spawn extension This may be the cause: Didn't get a frame from channel... I googled. It is recommended to disable busydetect, but no solution. Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sergio de los Santos ssantos @ hispasec.com Hispasec Sistemas S.L 902 161 025 29590 Málaga ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Didn't get a frame from channel
Using tdm400. While transfering a call from outside to another extensions, while this outside call is waiting with music, the another extension call hangs up suddenly, and the call is back to the outside call suddenly. Wathcing logs: Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462 bytes of audio while expecting 640 Jan 15 13:32:55 DEBUG[27850] channel.c: Didn't get a frame from channel: SIP/219-081d4d60 Jan 15 13:32:55 DEBUG[27850] channel.c: Bridge stops bridging channels SIP/219-081d4d60 and Zap/1-1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Hangup: channel: 1 index = 0, normal = 16, callwait = -1, thirdcall = -1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: disabled echo cancellation on channel 1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Updated conferencing on 1, with 0 conference users 15 13:32:55 VERBOSE[27850] logger.c: -- Hungup 'Zap/1-1' Jan 15 13:32:55 DEBUG[27850] app_dial.c: Exiting with DIALSTATUS=ANSWER. Jan 15 13:32:55 VERBOSE[27850] logger.c: == Spawn extension This may be the cause: Didn't get a frame from channel... I googled. It is recommended to disable busydetect, but no solution. Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration problem
firewall? i dont think so because sometimes the phone can register ok and sudendly the appears unregistered Leonardo Silva [EMAIL PROTECTED] ha escrito: 2006/10/31, Jon Farmer [EMAIL PROTECTED]: Sergio R. D'Ippolito wrote: Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information: */SIP/2.0 401 Unauthorized/* /Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-43bf8123;received=x.x.x.x/ /From: SPA922 sip:[EMAIL PROTECTED];tag=685bbad1fae3325do0/ /To: SPA922 sip:[EMAIL PROTECTED];tag=as4da6f6ce/ /Call-ID: [EMAIL PROTECTED]/ /CSeq: 5503 REGISTER/ /User-Agent: incore-PBX/ /Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY/ /WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=372b2479/ Asterisk is asking the phone to resend the registration with WWW-Authenticate using MD5 hash. Make sure the phone supports this and retry. Or you could turn this option off in the sip.conf. Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Maybe a Firewall ? -- Leonardo Silva fone: 16 8143-1146 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Registration problem
Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information: -- SIP read from x.x.x.x:1024: REGISTER sip:mysipserver.com SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-839856dc From: SPA922 sip:[EMAIL PROTECTED];tag=685bbad1fae3325do0 To: SPA922 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 5504 REGISTER Max-Forwards: 70 Contact: SPA922 sip:[EMAIL PROTECTED]:1025;expires=3600 User-Agent: Linksys/SPA942-4.1.12 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to x.x.x.x : 1025 (NAT) Transmitting (NAT) to x.x.x.x:1024: SIP/2.0 100 Trying Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-839856dc;received=x.x.x.x From: SPA922 sip:[EMAIL PROTECTED];tag=685bbad1fae3325do0 To: SPA922 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 5504 REGISTER User-Agent: incore-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-43bf8123;received=x.x.x.x From: SPA922 sip:[EMAIL PROTECTED];tag=685bbad1fae3325do0 To: SPA922 sip:[EMAIL PROTECTED];tag=as4da6f6ce Call-ID: [EMAIL PROTECTED] CSeq: 5503 REGISTER User-Agent: incore-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=372b2479 Content-Length: 0 Why the phone can not register? The password and username are ok. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] how to config chanspy
How can I do to select the channel to spy ? thanks De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Ralph Liebessohn Enviado el: Miércoles, 18 de Octubre de 2006 09:29 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] how to config chanspy On 10/17/06, Thirumal Saminathan [EMAIL PROTECTED] wrote: hi all, please any one help me ,how to configure chanspy application . and also send me if u have any sample configure file. -thiru Hi, It could be very simple, like: exten = 123,1,ChanSpy() ; Spy all channels or more accuracy: exten =124,1,ChanSpy(SIP) ; Spy all sip channels if I can help you more, let me know! -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Login user
Hi list! I have asterisk 1.2.12 installed and i need that the users can make a logon and logoff whit theirs phones on my asterisk pbx. Anybody know how can I do this ? Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] voicemailmain errors on CLI
You have to leave a message in the voicemail, then listen it and the error will not apear again. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Doug Lytle Enviado el: Miércoles, 13 de Septiembre de 2006 08:45 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] voicemailmain errors on CLI Benjamin Jacob wrote: Hello ppl, I am getting the following errors when accessing voicemails Sep 13 16:43:59 ERROR[19020]: app.c:1161 ast_lock_path: Unable to create lock file '/var/spool/asterisk/voicemail/pbx1VmBoxes/555123/Old': No such file or directory Just as the error states, the directory Old doesn't exist. Check to see if it does. If it is there, check it's permissions, if not then create it. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream and H.264 !
hi, I´ need some help to implement the Grandstream GXV-3000 in my * platform. Someone know the state of H.264 Video Codec for Asterrisk?? Thanks!!! p.D.: appreciate any help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Tycho Voicemail
Hi list! Im using Tycho software to see my voicemail, y can see de detail from the message but i cant hear de message. Somebody use that software any time ? have you the same problem ? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: problems with wevbmail
I could fix it. The problem was permissions on the directory /var/spool/asterisk/voicemail. Thanks De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Steven Enviado el: Miércoles, 23 de Agosto de 2006 08:01 a.m. Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] Re: problems with wevbmail Try running apache as the asterisk user instead of apache My assumption is that apache or your apache user does not have access to the voicemail folders. -- -- Steven http://www.glimasoutheast.org Sergio R. D'Ippolito [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]... I can login on the web http://myasterisk.com/cgi-bin/vmail.cgi without problems but i cant see the messages on any folder. Thanks, Sergio. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] NAT problems
Try changing the configuration on your PAP2 linksys, more precisly the part where is the NAT parameters, try changing the options from NO to YES. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de andrutto Enviado el: Miércoles, 23 de Agosto de 2006 03:41 p.m. Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] NAT problems Hi, Does anyone know how to solve this issue. I have Asterisk box on public IP and three clients connected to it. Unfortunately they are behind NAT (simple one-to-one). Those three clients can make outgoing calls hassle free, but when I try to make a call between them something is not right. I am using Linksys PAP-2 (two clients are connected to it) and one phone connected to planet VIP-156. When I try to make call between the phones connected to Linksys I am getting 488 Not Acceptable Here and when I try to reach the phone connected to planet I am getting silence after answer, but the phone can ring so I think that it is a RTP issue. I know that it is caused by the NAT, does anyone know how can I configure this to work appropriately. Cheers Andrutto -- Zostan Dziewczyna Lata! http://link.interia.pl/f1997 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Strange SIP response
I had the same problem. The problem was another sip extensions whit the same ip. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Rich Adamson Enviado el: Martes, 22 de Agosto de 2006 11:21 p.m. Para: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Strange SIP response Diego Andres Asenjo G. wrote: Hi, I am getting the following message on the CLI: -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I really need help with this error. Check the sip device config and make sure Do Not Disturb (DND), Call Forwarding, etc, have not be set. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problems with wevbmail
I can login on the web http://myasterisk.com/cgi-bin/vmail.cgi without problems but i cant see the messages on any folder. Thanks, Sergio. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Blog about asterisk and voip techology
Hi, The link in this mail is my blog about asterisk and voip technology: http://skalog.blogspot.com Thanks! Chiacchiera con i tuoi amici in tempo reale! http://it.yahoo.com/mail_it/foot/*http://it.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk technician needed in Buenos Aires Argentina
Dear guys:We are expanding our voip unit and currently looking for an Asterisk technician that can be part of our company here in Buenos Aires. If you know anyone who lives here and knows Linux and Asterisk, please contact me asap. Best regards,Sergio Veltriwww.pointhorizon.comSuipacha 119 Primer pisoCapital FederalBuenos Aires, Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting to a cluster of SIP servers
How about using LVS? http://www.ultramonkey.org/3/topologies/lb-overview.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: lunes, 24 de abril de 2006 17:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Connecting to a cluster of SIP servers You can't use round robin DNS. Round robin DNS will cause every SIP packet to potentially go through a different static path, which will break things. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Saturday, April 22, 2006 5:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Connecting to a cluster of SIP servers Although there maybe a better way, this would work: 1. Add the IP's into your sip.conf and set qualify=yes. 2. Make your dialplan something like the following: exten = _X.,1,Dial,SIP/[EMAIL PROTECTED] exten = _X.,2,Hangup exten = _X.,102,Dial,SIP/[EMAIL PROTECTED] exten = _X.,103,Hangup exten = _X.,203,Dial,SIP/[EMAIL PROTECTED] exten = _X.,204,Hangup exten = _X.,304,Dial,SIP/[EMAIL PROTECTED] exten = _X.,305,Hangup This would make your failover work but certainly wouldn't help with the load balancing between the servers. If any cannot qualify or are congested, they will automatically failover to the next server. I believe most people use an SER proxy for this type of application. It seems to work well with the round robin type DNS. William -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hill Sent: Saturday, April 22, 2006 5:13 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Connecting to a cluster of SIP servers My Asterisk server is connecting to sip.plus.net, which resolves to multiple IP addresses: sip.plus.net. 300 IN A 84.92.0.75 sip.plus.net. 300 IN A 84.92.0.76 sip.plus.net. 300 IN A 84.92.5.189 sip.plus.net. 300 IN A 84.92.5.190 If one of these machines is down (i.e. it's not replying to the SIP packets or it's sending back ICMP Port Unreachable), Asterisk keeps trying the same server. Shouldn't Asterisk move on to the next server automatically in this case? It seems to only way to do this at the moment is to run the reload command, which causes it to do a DNS lookup and it may then pick one of the other servers. -- - Steve xmpp:[EMAIL PROTECTED] sip:[EMAIL PROTECTED] http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. No confidentiality or privilege is waived or lost by any wrong transmission. If you have received this message in error, please immediately destroy it and kindly notify the sender by reply email. You must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message if you are not the intended recipient. Opinions, conclusions and other information in this message that do not relate to the official business of Ydilo Advanced Voice Solutions, S.A. shall be understood as neither given nor endorsed by it. -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 won't dial (sccp)
[EMAIL PROTECTED] ha scritto: On Wed, Apr 12, 2006 at 09:32:12PM +0200, Sergio Chersovani wrote: [EMAIL PROTECTED] ha scritto: context = from-sccp-intenal I guess intenal is not the righe context :-) Sergio The from-sccp-internal is almost an exact copy of my from-sip-internal context, which works fine there's a typo in your sccp.conf intenal instead internal, so of course the context does not exists Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 won't dial (sccp)
[EMAIL PROTECTED] ha scritto: context = from-sccp-intenal I guess intenal is not the righe context :-) Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation problem
Avi Miller ha scritto: Giuseppe wrote: Can anybody tell me if there is some error or something missing in this configuration please? I have the same card in a few of my servers and the echo canceller works just fine. I'm not 100% sure, but something does jump out at me: Mar 31 16:40:21 WARNING[29878]: chan_capi.c:3334 show_capi_conf_error: ISDN3: conf_error 0x300b PLCI=0x103 I guess you have to set the old echo facility number in your capi.conf Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SMS in Spain (it seems Protocol 2)
When I was in Telefonica I+D I developed an software for windows that allows
sending sms throw an ISDN line. It was more than 3 years ago and I don't recall
to many details but we had to implement ETSI ES 201 912 and
make an V28 modem emulation over ISDN.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carles Pina i
Estany
Sent: jueves, 30 de marzo de 2006 18:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SMS in Spain (it seems Protocol 2)
Hello,
On Mar/30/2006, Fran wrote:
I guess Protocol 1 is UBS1. I think it should be.
ok, me too...
No, i have never tested Asterisk sending messages.
We have tested some fixed devices (UBS1, UBS2 Domo type)
I have only checked Domo phone, but I don't know which protocol it is using.
Julian, from Asterisk-es (and he is here too) sent me some time ago this
link:
http://www.rtx.dk/Files/Filer/tekniske%20artikler/SMStransmissionwithinthePSTN.pdf
Maybe it is not updated, in topic about Protocol 1 and 2...
The UBS1 SMS Service is 900716800
Ok, I am using this one.
What error do u have? Timeouts? etc?
Well, I am doing this file:
Channel: Zap/1/900716800
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: smsdial
Priority: 1
Callerid: hola phone_of_FXO_card
Extension: phone_of_recipient
In extensions.conf I have this information:
[smsdial]
exten = _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME})
exten = _X.,2,SMS(${CALLERIDNUM})
exten = _X.,3,Hangup
(it is included from general section, etc.)
When I copy .call file to /var/spool/asterisk/outgoing, in Asterisk
console appears:
*CLI -- Attempting call on Zap/1/900716800 for [EMAIL PROTECTED]:1 (Retry
1)
Channel Zap/1-1 was answered.
-- Executing SMS(Zap/1-1, FXO_phone||phone_of_recipient|hola) in new
stack
-- Executing SMS(Zap/1-1, FXO_phone) in new stack
Mar 30 17:55:39 WARNING[11371]: chan_sip.c:9601 handle_response_register: Got
200 OK on REGISTER that isn't a register
== Spawn extension (smsdial, recipient_phone, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
Mar 30 17:56:27 NOTICE[11380]: pbx_spool.c:280 attempt_thread: Call completed
to Zap/1/900716800
If I change 900716800 phone to France SMSC phone (0033809101000), then
it appears:
*CLI -- Attempting call on Zap/1/0033809101000 for [EMAIL PROTECTED]:1
(Retry 1)
Channel Zap/1-1 was answered.
-- Executing SMS(Zap/1-1, from_phone||to_phone|hola) in new stack
-- Executing SMS(Zap/1-1, from_phone) in new stack
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
== Spawn extension (smsdial, 600512220, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
Mar 30 17:59:08 NOTICE[11403]: pbx_spool.c:280 attempt_thread: Call completed
to Zap/1/0033809101000
I rode that it should appear TX and RX lines (of course). SMS is not sent,
but maybe France SMSC is checking something (like I am not customer of there
:-) )
I don't have big knowledge about Asterisk. Maybe it is other stupid thing,
and not protocols issues...
--
Carles Pina i EstanyGPG id: 0x8CBDAE64
http://pinux.info Manresa - Barcelona
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If you have received this message in error, please immediately destroy it and
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You must not, directly or indirectly, use, disclose, distribute, print, or copy
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Ydilo Advanced Voice Solutions, S.A. shall be understood as neither given nor
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[Asterisk-Users] TDM11B desperate Help wanted
Hello:After configuring FXS and FXO channels of a TDM11B card, I can make calls from the telephone attached to the TDM11B card to the outsite (the PSTN, analog line), The problem is when I try to dial from the PSTN to my asterisk box (it has to make ring the handset on the TDM card), but I got infinite Poopy (00) on card 4! What is the problem?.This is the output of the wctdm module loaded with debug=1ACPI: PCI interrupt :00:08.0[A] - GSI 19 (level, low) - IRQ 169Freshmaker version: 73Freshmaker passed register test ProSLIC on module 0, product 0, version 2ProSLIC on module 0 seems sane.ProSLIC on module 0 powered up to -75 volts (c9) in 8 msLoop current set to 20mA!Post-leakage voltage: 33 voltsProSLIC on module 0 powered up to -72 volts (c2) in 5 ms Loop current set to 20mA!Calibration Vector Regs 98 - 107:98: 1199: 10100: 0e101: 0e102: 00103: 54104: 05105: 2a106: 20107: 08Init Indirect Registers completed successfully. Proslic module 0 loop current is 20mAModule 0: Installed -- AUTO FXS/DPOProSLIC on module 1, product 0, version 0Module 1: Not installedProSLIC on module 2, product 0, version 0Module 2: Not installed ProSLIC on module 3, product 0, version 0VoiceDAA System: 04ISO-Cap is now up, line side: 03 rev 03Module 3: Installed -- AUTO FXO (FCC mode)Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) 10299583 Polarity reversed (0 - -1)wctdm: Card 0 Going off hookwctdm: Card 0 Going on hookand when receiving a call from the PSTN I see this on /var/log/messagesMar 27 22:32:25 lineox kernel: Registered tone zone 2 (France) Mar 27 22:32:36 lineox kernel: RING on 1/4!Mar 27 22:32:36 lineox kernel: Poopy () on card 4!Mar 27 22:32:36 lineox kernel: Poopy (00) on card 4!Mar 27 22:32:44 lineox last message repeated 527 times What should be the problem?. What am I doing wrong?.my ztcfg -vvv saysZaptel Configuration==Channel map:Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 04: FXS Kewlstart (Default) (Slaves: 04)2 channels configured.Thanks a lot in advance for the help. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help: Using asterisk and mysql for a university project
Hello all, I want to use mysql for to save the users of my asterisk PBX. I use the realtime solution with mysql but when I made the sip show peers command doesnt appear my users. My configurations are: res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = asterisk dbpass = satec dbport = 3306 dbsock = /var/run/mysqld/mysqld.sock sip.conf [general] dbuser=asterisk dbpass=satec dbhost=127.0.0.1 dbname=asterisk table=sipusers rtcachefriends=yes extconfig.conf sipusers = mysql, asterisk, sipusers sippeers = mysql, asterisk, sipusers and the output of the realtime mysql status asterisk2006*CLI realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username asterisk for 3 minutes, 42 seconds. asterisk2006*CLI sip show peers Name/username Host Dyn Nat ACL Port Status 0 sip peers [0 online , 0 offline] asterisk2006*CLI sip show users Username Secret Accountcode Def.Context ACL NAT asterisk2006*CLI What is the problem? Thanks and Regards, Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk sip and radius authentication
Hello all, I am new in asterisk configuration. I want to configure a Radius server to authenticate the sip users of asterisk. I have trying to use the next document: http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html Can you help me? Regards, Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x
I have an [EMAIL PROTECTED] installed and then upgraded to 1.2.4 with bristuff and i have the same problem. If you see the logs the INSERT trace has wrong values before the comand is executed. By the way, everyone of us that have this problem use HFC cards? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: jueves, 09 de febrero de 2006 18:28 Para: Jeroen Zwarts; Asterisk Users Mailing List - Non-Commercial Discussion CC: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x You are in my same situation. I thought I solved the problem (if you look at tomorrow post) but it isn't My situation is a bit different: I have the last bristuffed version of asterisk 1.2.4 (released yesterday) And I also have 2 zaphfc cards. but the behaviour is absolutely the same If you restart asterisk, you get one or two calls ok, the again the problem On the first zaphfc, the problem is almost immediate (1 or two calls) the second is stronger, and is ok for a longer period ( 1 day ??) then it also falls in problem on clid and src It seems to me some buffer overwrite problem. the clid is trasmitted ok to the internal phones. So I am not alone on this side... Andrea Jeroen Zwarts [EMAIL PROTECTED] nl To Sent by: asterisk-users@lists.digium.com asterisk-users-bo cc [EMAIL PROTECTED] m.com Subject [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x 09/02/2006 11.05 Please respond to Jeroen Zwarts [EMAIL PROTECTED] nl; Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I have a problem with CDR recording in Asterisk 1.2.x. This is the situation: An Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1 with florz) machine with a single HFC-S ISDN BRI card. I log the call records to both the Master.csv and MySQL. The problem is that when an incoming call from the ISDN line is logged to the CDR, the src and the clid field show up as something like 'h? ' (random weird ASCII characters). This is in the MySQL table as well as the Master.csv, so my guess is that it is not a MySQL problem. Furthermore, I don't think it is a zaptel/bristuff problem, because my AGI scripts get the incoming number without problems all the time. The internal SIP calls are logged without a problem all the time. It's only ISDN calls from the outside world that are corrupt. When I stop Asterisk with stop now and restart it, the src and clid fields are OK for a while, but after a few calls, or as some time passes by (I don't know what triggers it), it goes back to the 'random ASCII weirdness'. I also tested this with Asterisk 1.2.4 (BRIstuffed-0.3.0-PRE-1h with florz) and I have the same problem. Again, when I start Asterisk, everything is OK for a while, and then suddenly, the src and clid fields are like 'ÀÜ' Anybody has a clue as where to start looking for a solution for this problem? I can't seem to find a single post, list e-mail or bug related to this problem. Thanks, Jeroen Zwarts ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users
Re: [Asterisk-Users] No audio? Update your Asterisk
Roger Hill wrote: I'm picking up the tail end of a thread, so apologies if this is offtrack... Have you perhaps got an old set of EXECUTABLES in your path, that are being picked up before your newly compiled ones? If you are under linux rm /usr/lib/asterisk/modules/* rm /usr/include/asterisk/* cd asterisk-1.2.4 make clean make upgrade asterisk -r stop now safe_asterisk that's all Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sccp availability?
Andy Webster ha scritto: hi, I'm trying to get the latest chan_sccp. The links from http://chan-sccp.berlios.de are all dead. Is it just me? Does anyone know an alternate source to get chan_sccp? Just tested, all the links work Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hangs on 1.2.1
Mark Johnson ha scritto: Feb 1 09:10:33 WARNING[5327]: channel.c:784 channel_find_locked: Avoided deadlock for '0xbf002d10', 10 retries! Yes, the chan_sccp could lock the asterisk channel. To fix it I need a sccp debug 10 log of the call that is locking the channel Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] changing cisco 7940/7960 standard menus ?
Mark Johnson ha scritto: I went though the same thing. I don't think you can change the menus. I simply set up Asterisk to Blind Xfer with the # key. So instead of using the softkeys, you hit # and then the extension and off the call goes. It works out nice because if you go to a different phone, the procedure stays the same. You can change the softkeys order editing the sccp_protcol.h from line 1060 Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Kirk IP600
Giordano Grandis ha scritto: I installed the chan_sccp and configured the sccp.conf, but when try to start asterisk I get this error [chan_sccp.so]Jan 31 10:31:15 WARNING[19727]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: ast_park_call Jan 31 10:31:15 WARNING[19727]: loader.c:391 load_modules: Loading module chan_sccp.so failed! you have to load the module res_features.so Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P FXS problem
Hi I'm getting strange problems with a fxs port of a didium TDM400 card. From time to time the telephone i've got plugged stop working correctly. I pickup the phone and i got dial tone, but if i try to dial it does nothing, as if it doesn't recognize de tones. If i press the flash key it works and any key pressed later wroks also. The only way of making it work is unplugging it and plugging it again till it works. Any ideas of what could be happening or at least which parameters in the zaptel driver could i start experimentin with? By the way, the phone if got plugged into is the standar phone that Telefonica sells here in Spain, so i would like to make asterisk work with that model. Greetings Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom 501 horrible echo
This is entirely SIP The behavior is only SIP to SIP...SIP to PSTN or PSTN to SIP = OK When one or both use speaker phone, the behavior is present. Both Handset or Headset = OK. How about trying with different codecs? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Draytek Vigor 2900 Asterisk
Jonathan Attwood wrote: I'm in conversation with Draytek's pre-sales dept.. I bought a 2600 2 years ago and I had alot of NAT problem, because the SPI was changing the externhost (sip.conf) ip address with the local private address forwarding the packets, so the audio stream was failing. I sent all the debug logs to the draytek dev team, but they were slow on updates to I bought a new and different brand router. Hope they fixed that issue in the new firmwares Good luck Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco phone issue
Scott DesBles wrote: I am working on adding three older Cisco phones to *, two 12SPs and one 30VIP. One of the 12SPs (griffin) and the 30VIP (scott) is booting correctly and I have dial tone. The other 12sp starts up, then I get a message on the display stating Requesting Try chan_sccp instead chan_skinny You have to play a bit with the configuration. Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hint on Zap channels
[EMAIL PROTECTED] ha scritto: is it possible to use the cid of a isdn-phone as well to identify multiple devices behind one line ? I did not understand the question, what you mean? Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hint on Zap channels
[EMAIL PROTECTED] ha scritto: has anyone an working example of a hint-entry with a Zap-Channel ? I've got hint working with SIP and SCCP but Zap doesn't seem to work Fixed in current CVS 1.2 and HEAD older versions have a case sensitivity issue so you have to write it in the right way this one works exten = 1, hint, Zap/1 this one does not work exten = 1, hint, ZAP/1 this one does not work exten = 1, hint, zap/1 Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 Reboot
Kristian Kielhofner ha scritto: Or you can keep using the phones with SIP and use sip_notify. I think Ciscos support it. In my last try it was not doing it on cisco sip phones. Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting Language
René Enskat [Teamware GmbH] ha scritto: -- Executing Set(SCCP/1000131-0006, Language()=de) edit your sccp.conf and in the general section set language=de; Default language setting Sergio Chersovani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Dynamic DNS
Branko Samardzic wrote: thought, it is DNS propagation problem, but it is NOT! Even one hour after IP change, machine A still points to old IP address and says that it is not reachable. I bet it is a DNS cache problem. Probably the machine A uses a cache dns and the record is not up to date. You have to run nslookup from the machine A to understand if the record was updated. Set the /etc/resolv.conf to point to a ISP dns server. Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Internet connection
José Luis Gómez ha scritto: Thanks, I will try thats. There was an issue in the ast_sip_ouraddrfor function. When the dns is down it fails to get the right address, you can easy patch it looking to the new code Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?
Noc Phibee ha scritto: it's possible to upgrade the firmware of a cisco 7910 with asterisk ? You need the legal firmware upgrade file download the chan_sccp code from http://chan-sccp.berlios.de configure it and use the imageversion param to upgradde the phone firmware. Of course you need a tftpserver and if you run a tftpserver you just need a SEPmac to upgrade the phone So the correct answer is: you don't need a CCM nor asterisk to upgrade a cisco phone firmware. You just need the firmware file, a tftpserver and a configuration file (SEPmac) take a look here http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?
Noc Phibee ha scritto: But cisco france say me that i cant' bye SmartNet contract on this product. why not? You can buy those smartnet contract via internet. You just need to mail US cisco and ask them for the contract activation. Only one solution are possible: Bye a special contract at $180.00 ... buy CON-SW-VPKG1 59 euros in europe Pff i can bye a new equipment with this price hihihi yep that is cisco i can't guest the latest firmware, for me i thinks that the solution are buy new voip phone and put the 7910 in Dead Yes you are right Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco Phone 7910
Noc Phibee ha scritto: i have buy a used Cisco Phone 7910 for use with my asterisk. The firmware version are 3.2(2.8), it's good for connect to asterisk ? That is an old firmware. atest is 5.07 Try http://chan-sccp.berlios.de for the sccp channel driver For update the fiormware, where i can get a new firmware ? You need a cisco contract. http://froogle.google.com/froogle?q=CON-SNT-CP7910 Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk fax
Tom Rymes ha scritto: However, I do think it is fairly clear that using an ATA is a less than ideal solution for any serious faxing, since the fax protocol often doesn't play nicely with the tendency of VOIP to occasionally lose packets. YMMV, though, so try it out Yep I can confirm this, I did try fax with TDM400 and ata devices (handytone and pap2) in a lan and the fax works when you disable the echo cancel and play a bit with the volume, but the retransmissions (or page cut off) are too much for an office use. So I'm back to an analog pstn solution for the fax and a isdn fax card solution with hylafax as software fax machine Maybe the t38 stuff will help in the future Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon Diva Server query
Erik Slooff ha scritto: I would like to suggest one small addition for clarity: you will *need* to have isdn4linux and capi4linux installed on your system in order to get chan_capi-cm installed. You just need the capi20 lib in order to use the chan_capi wget ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots/isdn4k-utils-CVS-2005-10-28.tar.bz2 tar xvjf isdn4k*bz2 cd isdn4* ./configure make make install that's all Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon Diva Server query
Ok, pay attention to /dev/capi20 device it must exists with the right permissions You just need the capi20 lib in order to use the chan_capi wget ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots/isdn4k-utils-CVS-2005-10-28.tar.bz2 tar xvjf isdn4k*bz2 cd isdn4* ./configure make make install that's all Sergio Great, that's clear for me now. Maybe a good idea to add this to the wiki page. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon Diva Server query
David Waugh ha scritto: I don't have isdn4linux and capi4linux installed but do have isdn4k-utils-devel-3.2-13.p1.1 isdn4k-utils-3.2-13.p1.1 Those are old packages, I suggest you to uninstall it and manual compile the version I posted in a previous release There are alot of changes in the newer versions Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and DrayTek Vigor2600VGi
richard Coco ha scritto: so the signalisation is ok. I have only problem with RTP packets (one way audio) The vigor firmware was really buggy about it. For example it was not working when externhost or externip param is set in the sip.conf file. I did notify the bug to the vigor dev team, but I don't know if they have fixed the problem yet, mine is gone really soon So an upgrade is of course necessary. Anyway you could understand the problem capping the sip packets with ethereal on both sides. I bet the ip address of the asterisk rtp box changes passing thru the vigor box and of course the device are not able to establish a right 2way audio session Let me know Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7905 sccp Hold and Message buttons
Francesco Angi ha scritto: Two simple questions about Cisco 7905 on Asterisk using chan_sccp. 1) using both sccp firmware 5.0 and 6.1 I cannot put a call in hold, because there's no Hold Button at all! Is there a way to configure The 7905 has an hard button for the hold stuff, the button is the one on the top of the button 1 element in SEPmac_address.cnf.xml, I also put it into sccp.conf, but pushing Message always dials 8500. vmnum = 123456 in the line section Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Multiple Line Appearance
Matt Hoskins ha scritto: I'm currently using asterisk 1.0.7 with cisco 7960 SIP 7.5 phones. I'd like to use this phone for a receptionist so that she can take calls for 4 other people. Is this possible? The SIP firmware does not support it. You have to use SCCP to do that Is there any way to do this with SIP and the 7960? I've seen the 7914 but then I'd have to use SCCP and I'm not sure if it is stable enough for production use. Well give it a chance :-) http://chan-sccp.berlios.de Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Multiple Line Appearance
Matt Hoskins ha scritto: Alright, I'm inspired. I'll give it a shot. Should I use the asterisk hint system or is line appearance done in the sccp config file seperately? Do you have a configuration example? the configuration example is in the package conf/sccp.conf or take a look at the site http://chan-sccp.org/ Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 18
Gervais de Montbrun ha scritto: **keepalive = 5 set the keepalive to 60 or more speeddial = 500,500,[EMAIL PROTECTED] that phone should not be able to display a hint status so speeddial = 500,500 This is what is displayed in the console when I try to call the 12SP from the ATA The log could be more verbose than this. Set debug = 10 in your sccp.conf or in the console sccp debug 10 You should see what is happening with your audio stream Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with=?ISO-8859-1?Q? g729_?= codec and ATA 1
Gervais de Montbrun ha scritto: **I did this in the console and the output is below. It does not seem to say much to me about audio. Dunno why, but the phone is not sending an open receive channel ack. In fact it does ot open the rtp media port so the channel don't know where to send (udp port) the rtp packets What firmware are you running? Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 186
Gervais de Montbrun ha scritto: I downloaded the chan_sccp as you suggested, but it does not seem to support my Cisco 12 SP+. I can see that it would support the ata, but if it doesn't support my other phone, then I need the skinny protocol and then can't use sccp... :-( the 12SP should work Do you know if I can get it to work with both my Cisco 12 SP+ and my ATA-186? Well you just need to change the default tcp port you can use chan_sccp on port 2000 and chan_skinny on port 2001 Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 186
Gervais de Montbrun ha scritto: Asterisk has been crashing like crazy since trying to run the latest RC-1 version and it seems to crash every time I try to use my Cordless phone. I have set the ATA to use the g729 codec that I purchased from Digium. Below is an example of a debug output from my console: You may want to try the http://chan-sccp.berlios.de code Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk connected with CAPI
richard Coco ha scritto: i'm trying to install a EICON DIVA 4BRI (on CentOS 4.1 2.6.9-22.0.1.EL) using latest package from Maybe you just need to check for the libcapi20 and /dev/capi20 device Anyway you can compile a fresh libcapi from here ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots/isdn4k-utils-CVS-2005-10-28.tar.bz2 tar xjf isdn4k-utils-CVS-2005-10-28.tar.bz2 cd isdn4k* ./configure make clean make install ldconfig Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Disconnect problem
Tryt the latest chan_sccp release http://chan-sccp.berlios.de nr k ha scritto: I have configured Asterisk call manager and i conneted 2 cisco ata 186 (SCCP).I make call between the ata's through Asterisk.the phones are perfectly registered with asterisk i am able to make calls but the call not disconnected after hangup and also i got an error msg RECEIVE MESSAGE TYPE UNKNOWN; 26 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SCCP support is making good progress
Chris Bagnall ha scritto: lower soft buttons hae labels like Pnbsp;, and apart from the single This is a old firmware issue, upgrading the phone firmware everything is working ok with the 7960 Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
