Thyda ENG
2015-Jul-08 08:24 UTC
[asterisk-users] How to enable IM over the asterisk server
I just get started with it so my question maybe not well catch. Anyway to do the VOIP call and IM we need to use two difference servers? which one is asterisk for VOIP ? and other one for IM that is openfire ? or we can have other choice better than this ? Thank you for your help, I am waiting for your reply. Thyda On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland < kvandenouweland at vangenechten.com> wrote:> Hi Thyda, > > I think you should see these as two individual systems. (I'm not an expert > so just thinking out loud). > > Since you mention that you did a SIP mapping on Openfire, may I assume > that you have the Asterisk IM plugin? > > In case of yes: > Yes, there is a plugin between OpenFire and Asterisk but it is not > actively developed anymore since 2006 > http://www.igniterealtime.org/projects/asterisk/ > > So I don't think the plugin is really realiable anymore on current > versions. > > -- > > I consider them as 2 separate systems which have to work on their own. > Unfortunatly this means that every softphone has 2 accounts: one is SIP to > Asterisk, one is XMPP to Openfire. > > That way our users are able to call internal/external using Asterisk, but > do IM and internal calling via Openfire. (They can choose which source they > take) > > Openfire is connected to our AD so our users just can logon with their > Windows credentials. > > Unfortunatly, if you want a real production connection between Asterisk > and Openfire, I'm unable to assist since I don't have the knowledge of it. > sorry > > Hope this helps a bit. > kristof > >>> Thyda ENG <engthyda at gmail.com> 7/07/2015 11:28 >>> > Actually, I am using the openfire and I create two users with the SIP > mapping on the openfire to the asterisk server. I can register one user > with the openfire client(Spark) and yes it is connect to asterisk SIP also. > But with the other one user, I register it with the SIP client(Zoiper/ or > Linphone) and then I can make the call over these two SIP but they cannot > reach the chat. I wonder what should I config between openfire and asterisk > to enable chat over these two sip clients ? > I am waiting for your reply, Thank. > > Thyda > > On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland < > kvandenouweland at vangenechten.com> wrote: > >> Good morning Thyda; >> >> Perhaps somebody has a solution for using it on Asterisk itself but after >> some trying I added the Openfire server as a IM server. >> >> I was a bit afraid that 'if' I got it working properly we had to maintain >> it and off course had to troubleshoot it in case it didn't work anymore. >> >> I've read something that you add a ams_msg context in extensions.conf but >> that didn't work for me unfortunaly. It did work for SIP Messages on phones >> but not for IM. >> >> I found Openfire easier to configure and it added a full integration with >> our LDAP which allowed single sign so that users could use the same >> password and log on automatically with the Jitsi client. >> >> But if you have some specific questions, I will be glad to answer. >> >> //Kristof >> >>> Thyda ENG <engthyda at gmail.com> 7/07/2015 6:07 >>> >> I am currently, I create the VOIP server which enable the user to make >> the call over the asterisk server, Additionally now I want the user to be >> able to chat to each other too. >> I found some suggestion of using the openfire with asterisk but not much >> said on it, Anyway could you please share me how can I config the IM server >> over asterisk? >> >> I am waiting for your reply, >> >> Thyda >> >> -- >> This message has been scanned for viruses and dangerous content by >> *Cisa Antispam Service*, and is believed to be clean. >> >> >> Privileged Confidential Information may be contained in this message. If >> you are not the addressee indicated in this message (or responsible for >> delivery of the message to such person), you may not copy or deliver this >> message to anyone. >> In such case, you should destroy this message and kindly notify the >> sender by reply email. >> Please advise immediately if you or your employer does not consent to >> Internet email for messages of this kind. >> Opinions, conclusions and other information in this message that do not >> relate to the official business of my firm shall be understood as neither >> given nor endorsed by it. >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > > Privileged Confidential Information may be contained in this message. If > you are not the addressee indicated in this message (or responsible for > delivery of the message to such person), you may not copy or deliver this > message to anyone. > In such case, you should destroy this message and kindly notify the sender > by reply email. > Please advise immediately if you or your employer does not consent to > Internet email for messages of this kind. > Opinions, conclusions and other information in this message that do not > relate to the official business of my firm shall be understood as neither > given nor endorsed by it. > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150708/b8ca064d/attachment-0001.html>
Kristof Van Den Ouweland
2015-Jul-08 11:49 UTC
[asterisk-users] How to enable IM over the asterisk server
Hi, I think so yes unless somebody else can provide a better solution. (Perhaps I'm doing it wrong ;-) ) We have 2 asterisk servers (Xivo distribution based on Debian) whom work in Active/Passive cluster mode. Then we have a third server which is the OpenFire server (based on Ubuntu 14) So yes, we have 2 different servers for that. nb. The 3CX version of Asterisk (Windows Version) has a baked-in IM server so then you only need one. But it does require you to buy a license which I didn't want to because Asterisk is opensource. //Kristof>>> Thyda ENG <engthyda at gmail.com> 8/07/2015 10:24 >>>I just get started with it so my question maybe not well catch. Anyway to do the VOIP call and IM we need to use two difference servers? which one is asterisk for VOIP ? and other one for IM that is openfire ? or we can have other choice better than this ? Thank you for your help, I am waiting for your reply. Thyda On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland <kvandenouweland at vangenechten.com> wrote: Hi Thyda, I think you should see these as two individual systems. (I'm not an expert so just thinking out loud). Since you mention that you did a SIP mapping on Openfire, may I assume that you have the Asterisk IM plugin? In case of yes: Yes, there is a plugin between OpenFire and Asterisk but it is not actively developed anymore since 2006 http://www.igniterealtime.org/projects/asterisk/ So I don't think the plugin is really realiable anymore on current versions. -- I consider them as 2 separate systems which have to work on their own. Unfortunatly this means that every softphone has 2 accounts: one is SIP to Asterisk, one is XMPP to Openfire. That way our users are able to call internal/external using Asterisk, but do IM and internal calling via Openfire. (They can choose which source they take) Openfire is connected to our AD so our users just can logon with their Windows credentials. Unfortunatly, if you want a real production connection between Asterisk and Openfire, I'm unable to assist since I don't have the knowledge of it. sorry Hope this helps a bit. kristof>>> Thyda ENG <engthyda at gmail.com> 7/07/2015 11:28 >>>Actually, I am using the openfire and I create two users with the SIP mapping on the openfire to the asterisk server. I can register one user with the openfire client(Spark) and yes it is connect to asterisk SIP also. But with the other one user, I register it with the SIP client(Zoiper/ or Linphone) and then I can make the call over these two SIP but they cannot reach the chat. I wonder what should I config between openfire and asterisk to enable chat over these two sip clients ? I am waiting for your reply, Thank. Thyda On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland <kvandenouweland at vangenechten.com> wrote: Good morning Thyda; Perhaps somebody has a solution for using it on Asterisk itself but after some trying I added the Openfire server as a IM server. I was a bit afraid that 'if' I got it working properly we had to maintain it and off course had to troubleshoot it in case it didn't work anymore. I've read something that you add a ams_msg context in extensions.conf but that didn't work for me unfortunaly. It did work for SIP Messages on phones but not for IM. I found Openfire easier to configure and it added a full integration with our LDAP which allowed single sign so that users could use the same password and log on automatically with the Jitsi client. But if you have some specific questions, I will be glad to answer. //Kristof>>> Thyda ENG <engthyda at gmail.com> 7/07/2015 6:07 >>>I am currently, I create the VOIP server which enable the user to make the call over the asterisk server, Additionally now I want the user to be able to chat to each other too. I found some suggestion of using the openfire with asterisk but not much said on it, Anyway could you please share me how can I config the IM server over asterisk? I am waiting for your reply, Thyda -- This message has been scanned for viruses and dangerous content by Cisa Antispam Service, and is believed to be clean. Privileged Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not copy or deliver this message to anyone. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer does not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of my firm shall be understood as neither given nor endorsed by it. -- _____________________________________________________________________ -- 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 Privileged Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not copy or deliver this message to anyone. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer does not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of my firm shall be understood as neither given nor endorsed by it. -- _____________________________________________________________________ -- 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 Privileged Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not copy or deliver this message to anyone. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer does not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of my firm shall be understood as neither given nor endorsed by it. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150708/d2bb1cb3/attachment.html>
James Cass
2015-Jul-08 11:55 UTC
[asterisk-users] How to enable IM over the asterisk server
You can have the openfire server installed on the same server as asterisk without any issue, just size your server appropriately. Just keep in mind they are different services. James Cass <http://goog_987864563> jcass78 at gmail.com On Wed, Jul 8, 2015 at 4:24 AM, Thyda ENG <engthyda at gmail.com> wrote:> I just get started with it so my question maybe not well catch. Anyway to > do the VOIP call and IM we need to use two difference servers? which one is > asterisk for VOIP ? and other one for IM that is openfire ? or we can have > other choice better than this ? > Thank you for your help, I am waiting for your reply. > > Thyda > > > On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland < > kvandenouweland at vangenechten.com> wrote: > >> Hi Thyda, >> >> I think you should see these as two individual systems. (I'm not an >> expert so just thinking out loud). >> >> Since you mention that you did a SIP mapping on Openfire, may I assume >> that you have the Asterisk IM plugin? >> >> In case of yes: >> Yes, there is a plugin between OpenFire and Asterisk but it is not >> actively developed anymore since 2006 >> http://www.igniterealtime.org/projects/asterisk/ >> >> So I don't think the plugin is really realiable anymore on current >> versions. >> >> -- >> >> I consider them as 2 separate systems which have to work on their own. >> Unfortunatly this means that every softphone has 2 accounts: one is SIP to >> Asterisk, one is XMPP to Openfire. >> >> That way our users are able to call internal/external using Asterisk, but >> do IM and internal calling via Openfire. (They can choose which source they >> take) >> >> Openfire is connected to our AD so our users just can logon with their >> Windows credentials. >> >> Unfortunatly, if you want a real production connection between Asterisk >> and Openfire, I'm unable to assist since I don't have the knowledge of it. >> sorry >> >> Hope this helps a bit. >> kristof >> >>> Thyda ENG <engthyda at gmail.com> 7/07/2015 11:28 >>> >> Actually, I am using the openfire and I create two users with the SIP >> mapping on the openfire to the asterisk server. I can register one user >> with the openfire client(Spark) and yes it is connect to asterisk SIP also. >> But with the other one user, I register it with the SIP client(Zoiper/ or >> Linphone) and then I can make the call over these two SIP but they cannot >> reach the chat. I wonder what should I config between openfire and asterisk >> to enable chat over these two sip clients ? >> I am waiting for your reply, Thank. >> >> Thyda >> >> On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland < >> kvandenouweland at vangenechten.com> wrote: >> >>> Good morning Thyda; >>> >>> Perhaps somebody has a solution for using it on Asterisk itself but >>> after some trying I added the Openfire server as a IM server. >>> >>> I was a bit afraid that 'if' I got it working properly we had to >>> maintain it and off course had to troubleshoot it in case it didn't work >>> anymore. >>> >>> I've read something that you add a ams_msg context in extensions.conf >>> but that didn't work for me unfortunaly. It did work for SIP Messages on >>> phones but not for IM. >>> >>> I found Openfire easier to configure and it added a full integration >>> with our LDAP which allowed single sign so that users could use the same >>> password and log on automatically with the Jitsi client. >>> >>> But if you have some specific questions, I will be glad to answer. >>> >>> //Kristof >>> >>> Thyda ENG <engthyda at gmail.com> 7/07/2015 6:07 >>> >>> I am currently, I create the VOIP server which enable the user to make >>> the call over the asterisk server, Additionally now I want the user to be >>> able to chat to each other too. >>> I found some suggestion of using the openfire with asterisk but not much >>> said on it, Anyway could you please share me how can I config the IM server >>> over asterisk? >>> >>> I am waiting for your reply, >>> >>> Thyda >>> >>> -- >>> This message has been scanned for viruses and dangerous content by >>> *Cisa Antispam Service*, and is believed to be clean. >>> >>> >>> Privileged Confidential Information may be contained in this message. If >>> you are not the addressee indicated in this message (or responsible for >>> delivery of the message to such person), you may not copy or deliver this >>> message to anyone. >>> In such case, you should destroy this message and kindly notify the >>> sender by reply email. >>> Please advise immediately if you or your employer does not consent to >>> Internet email for messages of this kind. >>> Opinions, conclusions and other information in this message that do not >>> relate to the official business of my firm shall be understood as neither >>> given nor endorsed by it. >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 >>> >> >> >> >> Privileged Confidential Information may be contained in this message. If >> you are not the addressee indicated in this message (or responsible for >> delivery of the message to such person), you may not copy or deliver this >> message to anyone. >> In such case, you should destroy this message and kindly notify the >> sender by reply email. >> Please advise immediately if you or your employer does not consent to >> Internet email for messages of this kind. >> Opinions, conclusions and other information in this message that do not >> relate to the official business of my firm shall be understood as neither >> given nor endorsed by it. >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150708/a837bc65/attachment.html>
Thyda ENG
2015-Jul-08 15:45 UTC
[asterisk-users] How to enable IM over the asterisk server
Yes, I have though of setting them up on the same server(openfire, and asterisk) and the problem come in mind that how can register the user to openfire automatically when I register the user SIP on the asterisk server ? Do you have any idea? I am waiting for your reply. Thank, Thyda On Wed, Jul 8, 2015 at 6:55 PM, James Cass <jcass78 at gmail.com> wrote:> You can have the openfire server installed on the same server as asterisk > without any issue, just size your server appropriately. Just keep in mind > they are different services. > > James Cass <http://goog_987864563> > jcass78 at gmail.com > > > On Wed, Jul 8, 2015 at 4:24 AM, Thyda ENG <engthyda at gmail.com> wrote: > >> I just get started with it so my question maybe not well catch. Anyway to >> do the VOIP call and IM we need to use two difference servers? which one is >> asterisk for VOIP ? and other one for IM that is openfire ? or we can have >> other choice better than this ? >> Thank you for your help, I am waiting for your reply. >> >> Thyda >> >> >> On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland < >> kvandenouweland at vangenechten.com> wrote: >> >>> Hi Thyda, >>> >>> I think you should see these as two individual systems. (I'm not an >>> expert so just thinking out loud). >>> >>> Since you mention that you did a SIP mapping on Openfire, may I assume >>> that you have the Asterisk IM plugin? >>> >>> In case of yes: >>> Yes, there is a plugin between OpenFire and Asterisk but it is not >>> actively developed anymore since 2006 >>> http://www.igniterealtime.org/projects/asterisk/ >>> >>> So I don't think the plugin is really realiable anymore on current >>> versions. >>> >>> -- >>> >>> I consider them as 2 separate systems which have to work on their own. >>> Unfortunatly this means that every softphone has 2 accounts: one is SIP to >>> Asterisk, one is XMPP to Openfire. >>> >>> That way our users are able to call internal/external using Asterisk, >>> but do IM and internal calling via Openfire. (They can choose which source >>> they take) >>> >>> Openfire is connected to our AD so our users just can logon with their >>> Windows credentials. >>> >>> Unfortunatly, if you want a real production connection between Asterisk >>> and Openfire, I'm unable to assist since I don't have the knowledge of it. >>> sorry >>> >>> Hope this helps a bit. >>> kristof >>> >>> Thyda ENG <engthyda at gmail.com> 7/07/2015 11:28 >>> >>> Actually, I am using the openfire and I create two users with the SIP >>> mapping on the openfire to the asterisk server. I can register one user >>> with the openfire client(Spark) and yes it is connect to asterisk SIP also. >>> But with the other one user, I register it with the SIP client(Zoiper/ or >>> Linphone) and then I can make the call over these two SIP but they cannot >>> reach the chat. I wonder what should I config between openfire and asterisk >>> to enable chat over these two sip clients ? >>> I am waiting for your reply, Thank. >>> >>> Thyda >>> >>> On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland < >>> kvandenouweland at vangenechten.com> wrote: >>> >>>> Good morning Thyda; >>>> >>>> Perhaps somebody has a solution for using it on Asterisk itself but >>>> after some trying I added the Openfire server as a IM server. >>>> >>>> I was a bit afraid that 'if' I got it working properly we had to >>>> maintain it and off course had to troubleshoot it in case it didn't work >>>> anymore. >>>> >>>> I've read something that you add a ams_msg context in extensions.conf >>>> but that didn't work for me unfortunaly. It did work for SIP Messages on >>>> phones but not for IM. >>>> >>>> I found Openfire easier to configure and it added a full integration >>>> with our LDAP which allowed single sign so that users could use the same >>>> password and log on automatically with the Jitsi client. >>>> >>>> But if you have some specific questions, I will be glad to answer. >>>> >>>> //Kristof >>>> >>> Thyda ENG <engthyda at gmail.com> 7/07/2015 6:07 >>> >>>> I am currently, I create the VOIP server which enable the user to >>>> make the call over the asterisk server, Additionally now I want the user to >>>> be able to chat to each other too. >>>> I found some suggestion of using the openfire with asterisk but not >>>> much said on it, Anyway could you please share me how can I config the IM >>>> server over asterisk? >>>> >>>> I am waiting for your reply, >>>> >>>> Thyda >>>> >>>> -- >>>> This message has been scanned for viruses and dangerous content by >>>> *Cisa Antispam Service*, and is believed to be clean. >>>> >>>> >>>> Privileged Confidential Information may be contained in this message. >>>> If you are not the addressee indicated in this message (or responsible for >>>> delivery of the message to such person), you may not copy or deliver this >>>> message to anyone. >>>> In such case, you should destroy this message and kindly notify the >>>> sender by reply email. >>>> Please advise immediately if you or your employer does not consent to >>>> Internet email for messages of this kind. >>>> Opinions, conclusions and other information in this message that do not >>>> relate to the official business of my firm shall be understood as neither >>>> given nor endorsed by it. >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- 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 >>>> >>> >>> >>> >>> Privileged Confidential Information may be contained in this message. If >>> you are not the addressee indicated in this message (or responsible for >>> delivery of the message to such person), you may not copy or deliver this >>> message to anyone. >>> In such case, you should destroy this message and kindly notify the >>> sender by reply email. >>> Please advise immediately if you or your employer does not consent to >>> Internet email for messages of this kind. >>> Opinions, conclusions and other information in this message that do not >>> relate to the official business of my firm shall be understood as neither >>> given nor endorsed by it. >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 >>> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150708/20717241/attachment-0001.html>