Re: [asterisk-users] passing '302 moved temporarily' back to the SIP provider
Satish Barot gmail.com> writes:
>
> On 5/9/13, Carlos Alvarez televolve.com> wrote:
> > On Tue, May 7, 2013 at 10:05 PM, Satish Barot
> > gmail.com>wrote:
> >
> >>
> >>
> >> promiscredir= yes in sip.conf should help you achieve your
requirement.
> >>
> >
> > I haven't been able to get that to work in a similar situation,
except we
> > are the provider. It results in the new invite being from the CLID
of the
> > original caller, and fails.
> >
> >
> > --
> > Carlos Alvarez
> > TelEvolve
> > 602-889-3003
> >
> Completely misunderstood the OP!
> Revised solution:
> Set promiscredir= no in sip.conf. I assume you land your dids in
> [incoming-trunk] and here is the basic dialplan tested on 11 but
> should work on 1.8.
>
> [incoming-trunk]
> ;-- Handle Incoming DIDs. Mine start with 89 and are of 4 digits --;
> exten =>
_89XX,1,Noop(RDNIS=${CALLERID(rdnis)}::ANI=${CALLERID(ani)}::DNID=${CALL
ERID(dnid)})
> same => n,Set(__ORIGCHANNEL=${CHANNEL})
> same => n,Dial(SIP/${EXTEN},30)
>
> ;-- Dialplan to handle 302 Moved temporarily --;
> exten =>
_X..,1,Noop(ORIGCHANNEL=${ORIGCHANNEL}::RDNIS=${CALLERID(rdnis)}::ANI=${
CALLERID(ani)}::DNID=${CALLERID(dnid)}::CHANNELTYPE=${CHANNEL(channeltyp
e)})
> same => n,ExecIf($["${CALLERID(rdnis)}"!=""]?
ChannelRedirect(${ORIGCHANNEL},back2provider,${EXTEN},1)
> same => n,Hangup()
>
> [back2provider]
> ;--Send 302 back to provider --;
> exten => _X.,1,Transfer(${EXTEN})
> same => n,NoOp(TRANSFERSTATUS=${TRANSFERSTATUS})
> same => n,Hangup()
>
> --Satish Barot
> Ahmedabad, India
>
> --
> _
Hi Satish,
We want to configure following setup:
“A” initiated call to SIP1.
SIP1 redirected CALL to SIP2(first redirection.
SIP 2 return 302, and request a redirect to SIP3(with SIP3 IP in
return packet).
SIP1 receive a redirect from SIP2 with SIP3 IP.
SIP1 makes a call to SIP3.
SIP3 finally helps in landing a call to “B”
All SIP are asterisk servers.
Please help in configuring asterisk to send 302 request back to the
server SIP1.
We are not able to get anywhere.
Regards,
Abhishek
--
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Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file
Hi, There isn't an astirectory driver for Asterisk version 1.4. So I guess you'll have to use the asterisk realtime (res_config_ldap) driver. cheers Abhishek On 9/5/07, Alessandro Russo [EMAIL PROTECTED] wrote: Hi to all I'm using Asterisk 1.4 and I'd like to use LDAP instead of sip.conf... Now you suggest to use asterisk realtime (res_config_ldap) or astirectory?? Can I use one of them with version 1.4? thx On 8/30/07, Gavin Henry [EMAIL PROTECTED] wrote: No probs. On 29/08/2007, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Gavin, Thank you once again. Will have to talk it over with my prof before upgrading to Asterisk 1.4. The productive system is currently running on 1.2.6. Thanks Abhishek On 8/28/07, Gavin Henry [EMAIL PROTECTED] wrote: On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Gavin, Sorry for having miss pelt your name twice... Thank you once again for your prompt reply. Is this the correct version of the driver for Asterisk 1.2.x : res_config_ldap-v0.7.tar.gz from the link http://bugs.digium.com/view.php?id=5768 If you use an old version of res_config_ldap with Asterisk version 1.2, I'm pretty sure you'll be asked to upgrade to Asterisk 1.4 if you seek any help via the lists or bug tracker. If you can use the latest release of Asterisk, you should. Thank you for your time and patience, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin! ;-) As of today I am using the res_config_ldap of Astirectory in my test Asterisk system to connect to a test LDAP database of my University. Things seem to be working fine so far. Now I'm faced with the task of installing this in the productive system. Before doing so, I'd sure like to consider trying the RealTime database driver that you people have developed. Why so? because I trust your judgment. Thanks, but you should still test it yourself. I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. This would mean removing Astirectory module, installing the new driver and loading the new schema into LDAP. In my view, the latter part shouldn't be a concern because the old attributes and object classes (Astirectory) should in no way interfere with the new ones. Besides the old object classes could be deleted from LDAP. Also the former part shouldn't be of much concern either. Nope, you are correct. My only concern as of now is in the installation of the RealTime database driver because the 'readme' file does not say anything about the installation. It only says about the configuration after installation. From the link: http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/ Would it be sufficiant if I were to copy the makefile and res_config_ldap.c to the res/ directory of my running Asterisk and do make; make install? or do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk version is 1.2.6. This Digium version is for 1.4.x, not 1.2 Thanks in advance, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote: On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin ;-) Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. It's for any LDAP Compliant Directory Server. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. Not really. in the RealTime driver you can specify which LDAP attributes map to which Asterisk Config settings. The only reason why I preferred Astirectory over the LDAP real time driver
Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file
Dear Mr Gavin, Thank you once again. Will have to talk it over with my prof before upgrading to Asterisk 1.4. The productive system is currently running on 1.2.6. Thanks Abhishek On 8/28/07, Gavin Henry [EMAIL PROTECTED] wrote: On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Gavin, Sorry for having miss pelt your name twice... Thank you once again for your prompt reply. Is this the correct version of the driver for Asterisk 1.2.x : res_config_ldap-v0.7.tar.gz from the link http://bugs.digium.com/view.php?id=5768 If you use an old version of res_config_ldap with Asterisk version 1.2, I'm pretty sure you'll be asked to upgrade to Asterisk 1.4 if you seek any help via the lists or bug tracker. If you can use the latest release of Asterisk, you should. Thank you for your time and patience, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin! ;-) As of today I am using the res_config_ldap of Astirectory in my test Asterisk system to connect to a test LDAP database of my University. Things seem to be working fine so far. Now I'm faced with the task of installing this in the productive system. Before doing so, I'd sure like to consider trying the RealTime database driver that you people have developed. Why so? because I trust your judgment. Thanks, but you should still test it yourself. I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. This would mean removing Astirectory module, installing the new driver and loading the new schema into LDAP. In my view, the latter part shouldn't be a concern because the old attributes and object classes (Astirectory) should in no way interfere with the new ones. Besides the old object classes could be deleted from LDAP. Also the former part shouldn't be of much concern either. Nope, you are correct. My only concern as of now is in the installation of the RealTime database driver because the 'readme' file does not say anything about the installation. It only says about the configuration after installation. From the link: http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/ Would it be sufficiant if I were to copy the makefile and res_config_ldap.c to the res/ directory of my running Asterisk and do make; make install? or do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk version is 1.2.6. This Digium version is for 1.4.x, not 1.2 Thanks in advance, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote: On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin ;-) Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. It's for any LDAP Compliant Directory Server. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. Not really. in the RealTime driver you can specify which LDAP attributes map to which Asterisk Config settings. The only reason why I preferred Astirectory over the LDAP real time driver was the fact that there is no mapping required for SIP users and peers. OK, maybe I need to go and read more about Astirectory. Regards Abhishek On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote: Please see the official tracker in the Digium buglist: http://bugs.digium.com/view.php?id=5768 Here are the schemas we did for OpenLDAP: http://bugs.digium.com/file_download.php?file_id=14842type=bug http://bugs.digium.com/file_download.php?file_id=14841type=bug Also, for Novell eDirectory, see: http://forge.voicerd.org/frs/?group_id=7release_id=17 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users
Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file
Dear Mr Galvin, As of today I am using the res_config_ldap of Astirectory in my test Asterisk system to connect to a test LDAP database of my University. Things seem to be working fine so far. Now I'm faced with the task of installing this in the productive system. Before doing so, I'd sure like to consider trying the RealTime database driver that you people have developed. Why so? because I trust your judgment. I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. This would mean removing Astirectory module, installing the new driver and loading the new schema into LDAP. In my view, the latter part shouldn't be a concern because the old attributes and object classes (Astirectory) should in no way interfere with the new ones. Besides the old object classes could be deleted from LDAP. Also the former part shouldn't be of much concern either. My only concern as of now is in the installation of the RealTime database driver because the 'readme' file does not say anything about the installation. It only says about the configuration after installation. From the link: http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/ Would it be sufficiant if I were to copy the makefile and res_config_ldap.c to the res/ directory of my running Asterisk and do make; make install? or do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk version is 1.2.6. Thanks in advance, Abhishek * * * * On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote: On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin ;-) Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. It's for any LDAP Compliant Directory Server. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. Not really. in the RealTime driver you can specify which LDAP attributes map to which Asterisk Config settings. The only reason why I preferred Astirectory over the LDAP real time driver was the fact that there is no mapping required for SIP users and peers. OK, maybe I need to go and read more about Astirectory. Regards Abhishek On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote: Please see the official tracker in the Digium buglist: http://bugs.digium.com/view.php?id=5768 Here are the schemas we did for OpenLDAP: http://bugs.digium.com/file_download.php?file_id=14842type=bug http://bugs.digium.com/file_download.php?file_id=14841type=bug Also, for Novell eDirectory, see: http://forge.voicerd.org/frs/?group_id=7release_id=17 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ --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 ___ --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 -- http://www.suretecsystems.com/services/openldap/ -- http://www.suretecsystems.com/services/openldap/ ___ --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 ___ --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] No LongDistance for 1 Extension?
Dear all, I'm faced with a similar situation of segregating users in 3 different categories to be able to make: internal calls only (students); internal local calls (staff); and internal, local international calls (profs). I do understand that 3 different contexts would have to be defined in the extensions.conf file. Are there custom context modules for Asterisk 1.2.6version as well? If not, I'd really appreciate any suggestions or help in this regard. thanks, Abhishek On 8/27/07, Seysan [EMAIL PROTECTED] wrote: Thank you. Now that the conexts are different can all the extension call to echother ? Seysan On 8/27/07, Steven [EMAIL PROTECTED] wrote: This is what I did in Trixbox: I added this to extensions_custom.conf - [restrict-local-only] include = from-internal-additional-custom include = app-recordings include = app-callwaiting-cwoff include = app-callwaiting-cwon include = app-dialvm include = app-vmmain include = app-cf-busy-off include = app-cf-busy-off-any include = app-cf-busy-on include = app-cf-off include = app-cf-off-any include = app-cf-on include = app-cf-unavailable-off include = app-cf-unavailable-on include = ext-meetme include = app-calltrace include = app-directory include = app-echo-test include = app-speakextennum include = app-speakingclock include = app-dnd-off include = app-dnd-on include = app-pickup include = app-chanspy include = ext-test include = ext-local include = outrt-007-local-only include = restrict-invalid exten = h,1,Hangup [restrict-invalid] exten = _9.,1,Playback(feature-not-avail-line) exten = _9.,n,Playback(that-number) exten = _9.,n,Playback(is) exten = _9.,n,Playback(privacy-not) exten = _9.,n,Playback(accessible-through-system) exten = _9.,n,Busy() -- Then in trixbox, each extension has a context field. Change it from the default to restrict-local-only. Also, I added a Route called local-only This includes just our local exchanges, emergency, and toll free. Dial Patterns: 911 9|1248. 9|1576. 9|1713. 9|1800. 9|1810. 9|1866. 9|1877. 9|1888. I created the restrict-invalid context to play a recording when a call was blocked. It matches anything not specified in restrict-local-only or higher included contexts. This scenario work great for me. Supposedly there is a Trixbox module called CustomContexts http://aussievoip.com.au/wiki/freePBX-CustomContexts , but it is in beta and seems more complicated than my approach. It should be much more versatile, but I went with the quick fix. -- -- Steven http://www.glimasoutheast.org Thomas Kenyon [EMAIL PROTECTED] wrote in message news: [EMAIL PROTECTED] Seysan wrote: Hi all, I want to limit the outgoing trunk to certain extensions, so for example 6 extensions can call long distance, but 4 other extensions are not allowed to do so. How can I do it in FreePBX specially? I don't know about Trixbox per say, but normally you would have all the handsets that can make long distance calls in one context and all the ones that can't in another, then use dialplan logic to glue it all together. ___ --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 ___ --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 ___ --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 ___ --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] Authenticating SIP user in LDAP database instead of SIP.conf file
Dear Mr Gavin, Sorry for having miss pelt your name twice... Thank you once again for your prompt reply. Is this the correct version of the driver for Asterisk 1.2.x : res_config_ldap-v0.7.tar.gzhttp://bugs.digium.com/file_download.php?file_id=9565type=bug from the link http://bugs.digium.com/view.php?id=5768 Thank you for your time and patience, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin! ;-) As of today I am using the res_config_ldap of Astirectory in my test Asterisk system to connect to a test LDAP database of my University. Things seem to be working fine so far. Now I'm faced with the task of installing this in the productive system. Before doing so, I'd sure like to consider trying the RealTime database driver that you people have developed. Why so? because I trust your judgment. Thanks, but you should still test it yourself. I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. This would mean removing Astirectory module, installing the new driver and loading the new schema into LDAP. In my view, the latter part shouldn't be a concern because the old attributes and object classes (Astirectory) should in no way interfere with the new ones. Besides the old object classes could be deleted from LDAP. Also the former part shouldn't be of much concern either. Nope, you are correct. My only concern as of now is in the installation of the RealTime database driver because the 'readme' file does not say anything about the installation. It only says about the configuration after installation. From the link: http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/ Would it be sufficiant if I were to copy the makefile and res_config_ldap.c to the res/ directory of my running Asterisk and do make; make install? or do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk version is 1.2.6. This Digium version is for 1.4.x, not 1.2 Thanks in advance, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote: On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin ;-) Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. It's for any LDAP Compliant Directory Server. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. Not really. in the RealTime driver you can specify which LDAP attributes map to which Asterisk Config settings. The only reason why I preferred Astirectory over the LDAP real time driver was the fact that there is no mapping required for SIP users and peers. OK, maybe I need to go and read more about Astirectory. Regards Abhishek On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote: Please see the official tracker in the Digium buglist: http://bugs.digium.com/view.php?id=5768 Here are the schemas we did for OpenLDAP: http://bugs.digium.com/file_download.php?file_id=14842type=bug http://bugs.digium.com/file_download.php?file_id=14841type=bug Also, for Novell eDirectory, see: http://forge.voicerd.org/frs/?group_id=7release_id=17 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ --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 ___ --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 -- http://www.suretecsystems.com/services/openldap/ -- http://www.suretecsystems.com/services/openldap/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file
Dear Mr Galvin, Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. The only reason why I preferred Astirectory over the LDAP real time driver was the fact that there is no mapping required for SIP users and peers. Regards Abhishek On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote: Please see the official tracker in the Digium buglist: http://bugs.digium.com/view.php?id=5768 Here are the schemas we did for OpenLDAP: http://bugs.digium.com/file_download.php?file_id=14842type=bug http://bugs.digium.com/file_download.php?file_id=14841type=bug Also, for Novell eDirectory, see: http://forge.voicerd.org/frs/?group_id=7release_id=17 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ --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 ___ --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] Authenticating SIP user in LDAP database instead of SIP.conf file
Dear All, Am happy to say that I've successfully been able to register a SIP user from a soft phone terminal via LDAP. The biggest hurdle that I had to overcome was the LDAP-Asterisk schema. The schema example given in the astirectory installation document is incomplete. Here's are a few pointers in this regard: The attributes have to be defined in the following way. Also tab spaces should be avoided. dn: cn=schema changetype: modify add: attributetypes attributeTypes: ( 1.3.6.1.4.1.23935.5.4.1 NAME 'astUsername' DESC '' SUP name EQUALITY caseIgnoreMatch SUBSTR caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) NAME should be the same as objectIdentifier DESC should be the description of the attribute EQUALITY is the rule to use when doing a search/compare for an attribute value. SUBSTR is the rule to use when doing a substring search (*foo*) SYNTAX is the syntax (i.e., type) of the attribute. We should probably stick to syntaxes: 1.3.6.1.4.1.1466.115.121.1.15 - directoryString (UTF-8 string) 1.3.6.1.4.1.1466.115.121.1.26 - IA5String (ASCII String) 1.3.6.1.4.1.1466.115.121.1.27 - integer (Integer value) The object class has to be always defined as AUXILLARY and never ABSTRACT. dn: cn=schema changetype: modify add: objectclasses objectClasses: ( 1.3.6.1.4.1.23935.5.5.1 NAME 'astSipGeneric' DESC '' SUP top AUXILIARY MUST ( astContext ) MAY ( astSecret $ astPermit $ astDeny $ astMd5Secret $ astDtmfmode $ astCanreinvite $ astNat $ astCallgroup $ astPickupgroup $ astAllow $ astDisallow $ astInsecure $ astTrustrpid $ astProgressinband $ astPromiscredir $ astRegseconds $ astname $ astLanguage ) ) Best Regards Abhishek On 8/16/07, Anthony Francis [EMAIL PROTECTED] wrote: You will need to extend your schema to include all of the attributes that can be used in sip.conf plus the extra ones that allow realtime to store connection information. Please refer to the realtime info at voipinfo.org to get a feel for what your schema should look like. Anthony Abhishek M S wrote: Dear all, May I first introduce myself. I'm a student of HAW Hamburg University currently working for my professor on a VOIP project. We have a Debian Linux system (server) on which Asterisk 1.2.6 has been successfully installed and running. Also the asterisk SIP server has been connected to the PSTN so users could make calls externally. We use Xlite softphone to make calls between users in the network. Currently there are very few users and I have been able to register them in the in *sip.conf *file and declare extensions in the *extensions.conf *file. Now there is a requirement to assign extensions to all students in the university(over thousand) whose credentials and information is stored in the Novel based LDAP database. Moving along I've managed to successfully install astirectory which is a real time database driver that allows to fetch configuration data from LDAP directories. Have also installed the LDAPget module that can lookup data in the LDAP directory. I'm looking for SIP attributes on LDAP or an LDAP schema that would facilitate astirectory or LDAPget to retrieve the username, telephone number and password from the LDAP database to register the soft phone user. I'd be extremely grateful for any help or suggestion in this connection. Thanks in advance, Abhishek ___ --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 ___ --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 ___ --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
[asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file
Dear all, May I first introduce myself. I'm a student of HAW Hamburg University currently working for my professor on a VOIP project. We have a Debian Linux system (server) on which Asterisk 1.2.6 has been successfully installed and running. Also the asterisk SIP server has been connected to the PSTN so users could make calls externally. We use Xlite softphone to make calls between users in the network. Currently there are very few users and I have been able to register them in the in *sip.conf *file and declare extensions in the *extensions.conf *file. Now there is a requirement to assign extensions to all students in the university(over thousand) whose credentials and information is stored in the Novel based LDAP database. Moving along I've managed to successfully install astirectory which is a real time database driver that allows to fetch configuration data from LDAP directories. Have also installed the LDAPget module that can lookup data in the LDAP directory. I'm looking for SIP attributes on LDAP or an LDAP schema that would facilitate astirectory or LDAPget to retrieve the username, telephone number and password from the LDAP database to register the soft phone user. I'd be extremely grateful for any help or suggestion in this connection. Thanks in advance, Abhishek ___ --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] Authenticating SIP user in LDAP database instead of SIP.conf file
Dear Mr.Anthony, Thank you so much for your responce.. Did go through the link as I'd done in the past. Would really appreciate if you could give me more specific links that could show a schema examples listing SIP attributes for LDAP. Appreciate your time and patience thanx Abhishek On 8/16/07, Anthony Francis [EMAIL PROTECTED] wrote: You will need to extend your schema to include all of the attributes that can be used in sip.conf plus the extra ones that allow realtime to store connection information. Please refer to the realtime info at voipinfo.org to get a feel for what your schema should look like. Anthony Abhishek M S wrote: Dear all, May I first introduce myself. I'm a student of HAW Hamburg University currently working for my professor on a VOIP project. We have a Debian Linux system (server) on which Asterisk 1.2.6 has been successfully installed and running. Also the asterisk SIP server has been connected to the PSTN so users could make calls externally. We use Xlite softphone to make calls between users in the network. Currently there are very few users and I have been able to register them in the in *sip.conf *file and declare extensions in the *extensions.conf *file. Now there is a requirement to assign extensions to all students in the university(over thousand) whose credentials and information is stored in the Novel based LDAP database. Moving along I've managed to successfully install astirectory which is a real time database driver that allows to fetch configuration data from LDAP directories. Have also installed the LDAPget module that can lookup data in the LDAP directory. I'm looking for SIP attributes on LDAP or an LDAP schema that would facilitate astirectory or LDAPget to retrieve the username, telephone number and password from the LDAP database to register the soft phone user. I'd be extremely grateful for any help or suggestion in this connection. Thanks in advance, Abhishek ___ --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 ___ --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 ___ --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
[asterisk-users] abhishek invites you to join Zorpia
Title: Zorpia: Invitation Email abhishek has sent you an invitation. Hi , Your friend abhishek from India just invited you to his/her online photo albums and journals at Zorpia.com. See abhishek's friends: Manish Shiraz Najaf So what is Zorpia? It is an online community that allows you to upload unlimited amount of photos, write journals and make friends. We also have a variety of skins in store for you so that you can customize your homepage freely. Join now for free! This message was delivered with the abhishek's initiation. If you wish to discontinue receiving invitations from us, please click here: Block future notifications Copyright(c) 2003-2005 Zorpia.com. All Rights Reserved. ___ --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] Re: Asterisk-Users Digest, Vol 20, Issue 211
Sir I want to set up asterisk server for only voip. I have installed it but unable to configure the extensions and other things. Pl. 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
RE: [Asterisk-Users] Asterisk Registering with SER question
Hi ryan , The header you are suspecting does not contains the registration info. , it is actually the return path for the ACK which will get generated in response to this packet. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Ryan PagquilSent: Tuesday, January 31, 2006 7:31 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk Registering with SER questionHi,On asterisk console I enabled SIP debugging and I found out that asterisk is sending this:Reliably Transmitting:REGISTER sip:imydomain.com SIP/2.0Via: SIP/2.0/UDP :x.x.x.x:5060;branch=z9hG4bK69398d1aFrom: sip:[EMAIL PROTECTED];tag=as1d1a85bcTo: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 REGISTERUser-Agent: Asterisk PBXExpires: 120Contact: sip:[EMAIL PROTECTED] --registered on SER Contact column on location tableEvent: registrationContent-Length: 0so it means that Asterisk is sending that information, how can I correct this? It should be sip:[EMAIL PROTECTED] no sip:[EMAIL PROTECTED] .Thanks in advance,Ryan ___ --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] Auto dialing
Hi list, I am facing a problem in auto diailing through call files. When i try to dial having this in my test.call , Channel: SIP/[EMAIL PROTECTED] Callerid: 3301 MaxRetries: 2 RetryTime: 60 WaitTime: 30 # # Assuming that your local extensions are kept in the # context called [extensions] # Context: test_in Extension: 1235 Priority: 1 I receive : *CLI -- Attempting call on SIP/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 1) Channel SIP/sip_proxy-out-1ebf was never answered. Feb 1 10:06:21 WARNING[6019]: cdr.c:548 ast_cdr_disposition: Cause not handled Feb 1 10:06:21 NOTICE[6019]: pbx_spool.c:266 attempt_thread: Call failed to go through, reason 8 even when i can dial out manually through the same context(sip_proxy-out) in sip.conf. Can somebody help me out of this problem. Thanks Abhishek ___ --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 in auto dialing through call files
Hi , I got a problem for you people. Actually i want to dial automatically any no. through my asterisk (registered on a remote SIP proxy). I asm using only SIP channels for extensions and to dial out too. There are two call flows which can be configured in call files. 1. My SIP extension rings , and when i offhook that phone , a outside no. gets dialed on SIP, so that asterisk give it to remote proxy server.( This things is going nicely) 2. But now when i want that a outside call must be dialed first , and when the other party pick up phone , then only asterisk gives the call to one of my extension. Please help me out . Thanks in advance. Regards Abhishek ___ --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] (no subject)
Hi, This is test mail. ___ --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] To Terry regarding Job requirement
Hi terry , I am an indian. Can u entertain my request , if so , i can send you my resume ___ --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] Please help
I am facing problem in playing a wav or gsm file on asterisk. The error i get whenever i tried is *CLI -- Executing BackGround(SIP/1235-98f6, /etc/asterisk-1.2.0/sounds/vm-goodbye.gsm) in new stack Jan 13 20:08:57 WARNING[6181]: file.c:508 ast_openstream_full: File /etc/asterisk-1.2.0/sounds/vm-goodbye.gsm does not exist in any format Jan 13 20:08:57 WARNING[6181]: file.c:820 ast_streamfile: Unable to open /etc/asterisk-1.2.0/sounds/vm-goodbye.gsm (format ulaw): No such file or directory Jan 13 20:08:57 WARNING[6181]: pbx.c:5747 pbx_builtin_background: ast_streamfile failed on SIP/1235-98f6 for /etc/asterisk-1.2.0/sounds/vm-goodbye.gsm == Auto fallthrough, channel 'SIP/1235-98f6' status is 'UNKNOWN' -- Saved useragent ZyXEL P2000W VoIP Wi-Fi Phone for peer 1235 Can somebody help out ___ --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] Please help
Hi tom, Thanks for a very quick reply. But iam sure i am having these file in th especified location with all th epermissions on it. - Original Message - From: Tom Vile [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 13, 2006 6:35 AM Subject: Re: [Asterisk-Users] Please help look in /etc/asterisk-1.2.0/sounds/ and see if you have sounds in that directory. On 1/14/06, Abhishek [EMAIL PROTECTED] wrote: I am facing problem in playing a wav or gsm file on asterisk. The error i get whenever i tried is *CLI -- Executing BackGround(SIP/1235-98f6, /etc/asterisk-1.2.0/sounds/vm-goodbye.gsm) in new stack Jan 13 20:08:57 WARNING[6181]: file.c:508 ast_openstream_full: File /etc/asterisk-1.2.0/sounds/vm-goodbye.gsm does not exist in any format Jan 13 20:08:57 WARNING[6181]: file.c:820 ast_streamfile: Unable to open /etc/asterisk-1.2.0/sounds/vm-goodbye.gsm (format ulaw): No such file or directory Jan 13 20:08:57 WARNING[6181]: pbx.c:5747 pbx_builtin_background: ast_streamfile failed on SIP/1235-98f6 for /etc/asterisk-1.2.0/sounds/vm-goodbye.gsm == Auto fallthrough, channel 'SIP/1235-98f6' status is 'UNKNOWN' -- Saved useragent ZyXEL P2000W VoIP Wi-Fi Phone for peer 1235 Can somebody help out ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] Please help
Thanks a lot. I was using the extension after the file name. - Original Message - From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 13, 2006 7:12 AM Subject: Re: [Asterisk-Users] Please help just remember that in Background() Playback() and friends, you must specify the file name without extension. Also, all sound should be in the directory specified in asterisk.conf Regards On 1/14/06, Abhishek [EMAIL PROTECTED] wrote: Hi tom, Thanks for a very quick reply. But iam sure i am having these file in th especified location with all th epermissions on it. - Original Message - From: Tom Vile [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 13, 2006 6:35 AM Subject: Re: [Asterisk-Users] Please help look in /etc/asterisk-1.2.0/sounds/ and see if you have sounds in that directory. On 1/14/06, Abhishek [EMAIL PROTECTED] wrote: I am facing problem in playing a wav or gsm file on asterisk. The error i get whenever i tried is *CLI -- Executing BackGround(SIP/1235-98f6, /etc/asterisk-1.2.0/sounds/vm-goodbye.gsm) in new stack Jan 13 20:08:57 WARNING[6181]: file.c:508 ast_openstream_full: File /etc/asterisk-1.2.0/sounds/vm-goodbye.gsm does not exist in any format Jan 13 20:08:57 WARNING[6181]: file.c:820 ast_streamfile: Unable to open /etc/asterisk-1.2.0/sounds/vm-goodbye.gsm (format ulaw): No such file or directory Jan 13 20:08:57 WARNING[6181]: pbx.c:5747 pbx_builtin_background: ast_streamfile failed on SIP/1235-98f6 for /etc/asterisk-1.2.0/sounds/vm-goodbye.gsm == Auto fallthrough, channel 'SIP/1235-98f6' status is 'UNKNOWN' -- Saved useragent ZyXEL P2000W VoIP Wi-Fi Phone for peer 1235 Can somebody help out ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] Recommendations on a WiFi phone for *?
I have tried Zyxel P2000W , it works very fine. - Original Message - From: Asterisk-User [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 10, 2006 3:23 AM Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *? Has anyone tried out Hitachi IPC-5000 ? It looks nice and it's a bit exensive, but I would still like to hear how does it behave around Asterisk. Ivan ___ --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] Problem with Xlite free phone(Xten)
Hi, got this when dial to Xlite User agent registered as 1234. BUt it can dial out. -- Executing Dial(SIP/1235-e9b6, SIP/1234) in new stack -- Called 1234 -- Got SIP response 486 Busy back from 202.54.195.89 -- SIP/1234-7c6b is busy == Everyone is busy/congested at this time (1:1/0/0) == Auto fallthrough, channel 'SIP/1235-e9b6' status is 'BUSY' Configuration in sip.conf for 1234 is as [1234] type=friend host=dynamic context=default username=1234 secret=1234 ;regexten=1234 Abhishek ___ --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] (no subject)
Hi all,
I am testing my hands on asterisk , but got stuck. Let me tell you i am
only using its VOIP functionlities
I have registered the asterisk server at a remote proxy server. My clients
registered at asterisk server can make outgoing calls , but the calls made
from outside is not incoming to any extension.
I have written
user:[EMAIL PROTECTED]/1234
in sip.conf.
and 1234are defined as
[1234]
type=friend
host=dynamic
context=test_in
user=phone
regexten=1234
in extensions.conf i am using
[test_in]
exten= 1236,1,Dial(SIP/sandhu)
exten= 1235,1,Dial(SIP/1235)
exten= _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten= 1234,1,Dial(SIP/1234)
My clients are on Xlite softphone.
Can anybody help out ?/
Abhishek
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[Asterisk-Users] Calls not incoming to any extension from remote proxy server
Hi all,
I am testing my hands on asterisk , but got stuck. Let me tell you i am
only using its VOIP functionlities
I have registered the asterisk server at a remote proxy server. My clients
registered at asterisk server can make outgoing calls , but the calls made
from outside is not incoming to any extension.
I have written
user:[EMAIL PROTECTED]/1234
in sip.conf.
and 1234are defined as
[1234]
type=friend
host=dynamic
context=test_in
user=phone
regexten=1234
in extensions.conf i am using
[test_in]
exten= 1236,1,Dial(SIP/sandhu)
exten= 1235,1,Dial(SIP/1235)
exten= _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten= 1234,1,Dial(SIP/1234)
My clients are on Xlite softphone.
Can anybody help out ?/
Abhishek
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Re: [Asterisk-Users] Calls not incoming to any extension from remoteproxy server
Thanks a lot for the reply. But i am sucessfully getting registered on the
remote proxy, so that i am getting right outputs as u said. I think that is
why only i am able to route calls outside to remote proxy,
The problem is when i am writing
register = user:[EMAIL PROTECTED]/1234
, the outside calls are not coming to 1234 extension , which is a Xlite
client.
The files configuration are as
sip.conf
register = user:[EMAIL PROTECTED]/1234
[1234]
type=friend
host=dynamic
context=test_in
user=phone
regexten=1234
extensions.conf
[test_in]
exten= 1236,1,Dial(SIP/sandhu)
exten= 1235,1,Dial(SIP/1235)
exten= _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten= 1234,1,Dial(SIP/1234)
Abhishek
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, December 21, 2005 6:03 AM
Subject: Re: [Asterisk-Users] Calls not incoming to any extension from
remoteproxy server
On Thursday 22 December 2005 04:45, abhishek wrote:
Hi all,
I am testing my hands on asterisk , but got stuck. Let me tell you i
am
only using its VOIP functionlities
I have registered the asterisk server at a remote proxy server. My
clients registered at asterisk server can make outgoing calls , but the
calls made from outside is not incoming to any extension.
I have written
user:[EMAIL PROTECTED]/1234
register = user:[EMAIL PROTECTED]/1234
is it not?
And when you do sip show registry
you see
server*CLI sip show registry
HostUsername Refresh State
proxy-ip:5060user105 Registered
Hope that gines you a clue.
benchev
in sip.conf.
and 1234are defined as
[1234]
type=friend
host=dynamic
context=test_in
user=phone
regexten=1234
in extensions.conf i am using
[test_in]
exten= 1236,1,Dial(SIP/sandhu)
exten= 1235,1,Dial(SIP/1235)
exten= _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten= 1234,1,Dial(SIP/1234)
My clients are on Xlite softphone.
Can anybody help out ?/
Abhishek
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61
Sir I am a novice user and want to set up the asterix for only Voip as a project in my final yr. computer engineeering. Pl. help me to do so . I will be highly thankful Abhishek Gangal ___ --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] What do you name yours
mine, on the stars of saturn options: Dione, Rhea, Titan, Mimas, Enceladus, Tethys, Hyperion, Iapetus, and Phoebe Abhishek -- Drishti-Soft Solutions Pvt Ltd http://www.drishti-soft.com On 5/12/05, Christopher Stephens [EMAIL PROTECTED] wrote: Mine is called 'blacksun', as that's where it's colo'd. (idiocy in a naming convention, I know.) On Wed, 11 May 2005 19:55:36 -0700 (PDT), Matt Klein [EMAIL PROTECTED] said: Mine is named spike... On Thu, 12 May 2005, Paul Hales wrote: We bought one of those books on the worst cars ever made...every page has great names... PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Thursday, 12 May 2005 1:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Andrew Latham Subject: Re: [Asterisk-Users] What do you name yours On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote: Naming Conventions for Asterisk Hostnames, . For an internal historical reason all ours come from the legends of Robin Hood. I used to work with a bunch of Lord of the Rings readers and all the machine names came from there. It always makes a good light discussion point. -- Dave Cotton [EMAIL PROTECTED] ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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 ___ 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 ___ 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 ___ 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] how to disconnect a call manually
soft hangup channel name -Abhishek Drishti-Soft Solutions Pvt Ltd http://www.drishti-soft.com On 5/2/05, Asterisk guy [EMAIL PROTECTED] wrote: 1 after giving command oh323 show channels, i want to disconnect a call, is there any command to disconnect a call? 2 how asterisk kill a hung/dead call ? for most commercial softswitch, there are a setting for maximum duration for a call. they will hang up it l if its duration reachs the limit. ___ 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 ___ 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] How can I eveluate trailing numbers in extensions.conf?
Hi,
you can use StripMSD if you are sure of the number of dialled digits.
or you can use {$EXTEN:-1:1}, where {$EXTEN:a:b} means first b digits
starting from a, from the front or back depending whether a 0 or a 0.
additionally, can have your own variables, just look at
pbx_retrieve_variable in pbx.c
-Abhishek
Drishti Soft
www.drishti-soft.com
On Sun, 13 Mar 2005 10:01:27 +, Umar Sear [EMAIL PROTECTED] wrote:
Checkout
http://www.voip-info.org/wiki-Asterisk+variables
I believe that should have the answer for you.
furthermore assuming that your number is always going to be 12 digits.
exten = _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number.
Hope this helps.
Umar
On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz [EMAIL PROTECTED] wrote:
Hi,
this message seems to not have made it to the list the first time - sorry
if it did.
My SIP provider includes trailing numbers to my account just fine, like
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKc62e.644d2d75.0
From: Anonymous sip:[EMAIL PROTECTED];tag=as4b25d20f
Call-ID: [EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED];tag=as406f4254
CSeq: 102 ACK
User-Agent: sipgate ser
Content-Length: 0
where 498645342456 is my SIP account phone number that can be reached from
the outside just fine. My question is, how can I evaluate the trailing 2
in my extensions.conf? This would be ideal for direct dialing to an attached
phone, and not be restricted to a single digit.
asterisk-dev The full number is included in the SIP message but does *
keep it
somewhere internally so that one could maybe add another externally usable
* variable? I browsed the source code but could not find anything... TIA!
Ciao,
hm
--
Today is the first day of the rest of the mess
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[Asterisk-Users] Asterisk with mediant 2000 - facing problems
Hi, I have been using/working on asterisk for some time now and presently was trying to configure asterisk to work with digium cards. It worked fine with the fxo/fxs cards, but now i'm trying to get it working by interfacing it with mediant t1 port. no avail ... anyone out there got it working, what particular configuration used on mediant (isdn signalling, framing, coding etc ??) and/or what configuration on asterisk side ?? if anyone has it working, please send me the ini file for the mediant and the zaptel.conf etc. would be extremely thankful. Regards Abhishek ___ 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
[Asterisk-Users] GUI based.. or ??
Hi, I am Abhishek from India. I am have studying Cisco VOIP since a couple of months.Searching for Soft PBX somenthing like (Cisco Callmanager) i came accros this Asterisk. I have to provide a a solution to a clinet where he wants a connectivity between his 3 offices across the WAN with a very limited amount of budget.Since i am not aware abt this product much, but was able to foind out the features of the product and was satisfied also, So i just wanted to know from u ppl (since u ppl are expert in this) that : 1.)does this product has got a GUI interface .?? 2.)Can we integrate Cisco or any other H/w with this.? 3.)it looks like freeware..isnt it.? Please do let me the details abt the same.. I ll be really greatful Thanking You, Regards Abhishek Katta - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 8:45 PM Subject: Re: [Asterisk-Users] Echo problem update - POSSIBLE SOLUTION So are saying that T2240 will gurantee no echo issues? Did you get any echo issues with a different PC with the same cards and Pstn lines? snip No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but... What's common and not so common between these _very_ diverse boxes? Nope. the intent of that post was only to suggest that echo resolution varies by system, and has nothing to do with how fancy/speedy of a Compaq/Dell/HP/IBM/insert-your-favorite-box-here you might be considering or have available, or how much you spent for it. The T2240 with tdm-x100p cards in one US case does not have echo after the echotraining=800 implementation. Don't read anything more into it then just that. (The echotraining=800 was enough of a change for that exact system implementation to function well. The next one may not.) Some strong arguments have been made off-list the existing echo cancellation function is highly dependent upon interrupt latency, motherboard chipset in use, PCI controller, and/or other system-level items that might even include driver inefficiencies of the NIC card. Its way to early to pin the issue any closer, and might even involve more then one item. (Gary Mart is focusing on this and I'm sure he would appreciate any technical/programming help he can get. Now I wish I wouldn't have let those skills go years ago.) Swapping motherboards can impact echo but doing so does not address the root cause, only the symptoms. It would be nice to know XXX board works and YYY board does not, but the professional approach should focus on the underlying issue(s) and correcting/compensating for those, if possible. It could be something as simple as a linux installation default (eg, assuming 33mhz buss, choice of drivers), or as complex as rewriting how the cancellation algorithm functions in general. It is known that a lot of implementations don't have echo, and apparently those boxes are using internal system resources that fall within the tolerances of the existing cancellation routines AND those boxes have been correctly interfaced to their pstn. Why others don't needs to be identified, and unfortunately, is not a simple task. In the past eight months we've all listened to suggestions that include killing the system's GUI interface, don't share interrupts, reverse tip ring, etc, etc. However, it now _appears_ those were probably addressing the symptom and not the root cause. It's still most appropriate to ensure the pstn interfacing is implemented correctly including source of T1 sync, impedance matching, adjust gain parameters to reasonable levels, use of proper interface cards for your country's pstn standards, etc. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
