September 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Cleanup destination_number with country code
by Michael Nielsen 09 Sep '15

09 Sep '15

I would like to check if the called number contains a country code, if not then one should be added. I've got a variable in my user's XML file called country_code containing +44. This is my code so far, however, I think the last to conditions overlap: <condition field="destination_number" expression="^(\+\d{3,20})$"> <action application="export" data="dialed_extension=$1"/> </condition> <condition field="destination_number" expression="^00(\d{3,20})$"> <action application="export" data="dialed_extension=+$1"/> </condition> <condition field="destination_number" expression="^(\d{3,20})$"> <action application="export" data="dialed_extension=${country_code}$1"/> </condition> I would like to change 00 to +, and if no + is on the number then use country_code from the user. If it does contain a +, then just use the dialled number.

2 3

Add country code if it aren't dialled.
by Michael Nielsen 09 Sep '15

09 Sep '15

I would like my dial plan in FS to automatically add a country code (from the users variable "country_code") if no country code has been dialled. This is part of my dial plan: <condition field="destination_number" expression="^\+(\d{3,20})$"> <action application="export" data="dialed_extension=$1"/> Is it possible to automatically check if either + or 00 is dialled, if not then add + <user country_code>. If 00 is dialled then this should be replaced with a +.

2 2

Using dialplan routing from ESL?
by René Weiss 09 Sep '15

09 Sep '15

Hi, I have set up some simple dialplan rules like: - Numbers with 4 digits or shorter: internal calls - 5 digits or longer: use external sip gateway This works perfectly fine with our (soft-)phones. Now I would also like to use the same rules when I make calls from our software over the ESL, but it seems that I always have to tell the "originate" command which gateway to use. Is there a syntax for "originate" (or another command) which uses the dialplan rules to decide which gateway to use, or do I have to implement the dialplan rules a 2. time in the software? Thanks, René

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Announcement: FreeSWITCH on PPC, FreeScale, POWER platforms
by Larry Morley 09 Sep '15

09 Sep '15

Hello, We're working on porting FS to the POWER architecture, from PPC G4 to POWER 8, with specific hardware and software platforms to be determined by the level of interest expressed in any given one. We've strongly advocated the use of PPC/POWER processors since early on, for both general purpose computing and in specific problem domains. The decision by IBM, Tyan, and other industry leaders to undertake the OpenPOWER initiative, combined with the recent availability of a POWER 8 system from Tyan [http://www.tyan.com/campaign/openpower/index.html] has reaffirmed our commitment to, and strengthened our belief in, the potential of this architecture, and how outstanding it is from any number of perspectives - including cost vs. benefit. Currently, we're testing a build of a port of the latest release of FreeSWITCH using G4 and G5 based systems capable of running Debian "Jessie". We've been careful to maintain the structure of the supporting libraries, building them only as .deb packages. Feature test results so far have been better than promising; and, we have no reason to believe the results of the remaining tests will be any less so. If you're interested in development of FS on a PPC / POWER platform, or would like more information on the current status of our project, please contact me. Thanks, Larry Larry Morley R&D Director Enlightened Computer Solutions, L.L.C. lmorley(a)neny.cslimits.net TEL: (442) 333 3416 FAX: (518) 284 0373

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Condition on if user exists.
by Michael Nielsen 09 Sep '15

09 Sep '15

I would like to the following: <condition IF-USER-EXISTS> </condition> <condition IF-USER-DO-NOT-EXISTS> </condition> I'm not able to check on regexp, as my user ID's match +44-12345678... And I would like to do one thing if the user exists and otherwise route to my PSTN gateway. I've tried to just use bridge with both local routing and PSTN calls, but then I try to call my PSTN everytime I call local users.

2 2

Multiple Conditions
by Tim King 09 Sep '15

09 Sep '15

I need to route calls to different destinations based on network ACL. I still need the calls to match a certain dial pattern how can I accomplish this? *All calls will match this first condition* <condition field="destination_number" expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$"> *If this* <condition field="${acl(${network_addr} list1)}" expression="true"/> *execute these actions* *if this* <condition field="${acl(${network_addr} list2)}" expression="true"/> *execute these actions*

2 1

Hardware Recommendations
by Adrian Thompson 09 Sep '15

09 Sep '15

Hello, We are new to FreeSwitch and would like to set up a PBX system with a BigBlueButton on the side. I've played around with BigBlueButton and have it working pretty flawlessly. - We would like to have two PBX (Fusion PBX GUI) systems at two sites with three remote sites piggy backing through VPN. - 10 Phone Numbers, 50 Extentions - Fail over to second site if first site is down - We have 4 Fax lines - We are looking for phones that can set presence (Available, Away, DnD, Custom) and have good quality sound - We should be able to support at least 3 lines per phone, but if a two line model with the above specs is available, we can work with that also I have looked over the hardware list on the FreeSwitch site, but I don't have too many resources to test different phones before implementation. The hardware I have looked at so far is the Polycom VVX 300 and The VVX 400 with expansion (Switchboard) but it looks like the DnD button doesn't really work as intended. What about Cisco Phones? I'd like to keep things under $100 USD for the IP Phone price per unit. SITE-1(Main)<---------->SITE-2(Fail-over) ^ | REMOTE-SITES-VPN Thank you in advance for any advice on the matter Adrian

3 2

Hardware Recommendations
by Adrian Thompson 08 Sep '15

08 Sep '15

Hello, We are new to FreeSwitch and would like to set up a PBX system with a BigBlueButton on the side. I've played around with BigBlueButton and have it working pretty flawlessly. - We would like to have two PBX (Fusion PBX GUI) systems at two sites with three remote sites piggy backing through VPN. - 10 Phone Numbers, 50 Extentions - Fail over to second site if first site is down - We have 4 Fax lines - We are looking for phones that can set presence (Available, Away, DnD, Custom) and have good quality sound - We should be able to support at least 3 lines per phone, but if a two line model with the above specs is available, we can work with that also I have looked over the hardware list on the FreeSwitch site, but I don't have too many resources to test different phones before implementation. The hardware I have looked at so far is the Polycom VVX 300 and The VVX 400 with expansion (Switchboard) but it looks like the DnD button doesn't really work as intended. What about Cisco Phones? I'd like to keep things under $100 USD for the IP Phone price per unit. SITE-1(Main)<---------->SITE-2(Fail-over) ^ | REMOTE-SITES-VPN Thank you in advance for any advice on the matter Adrian

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Re: [Freeswitch-users] OneWay audio after hold->unhold
by kaleem rehman 08 Sep '15

08 Sep '15

Hi Prashanth, is there any reason you are going for SRTP? I always advise customers to first setup normal link and fully test then move to SRTP. when you are saying freeswitch not responding with key, are you seeing the message in FS Debug being generated? if not then you will need to raise a JIRA. if you can, then next step would be to see this in traceSM logs and see if SM100 is getting these and how is SM responding to these and if sending to CM or not. thanks, kaleem On Tue, Sep 8, 2015 at 11:08 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: OneWay audio after hold->unhold (Prashanth Devarajappa) > > > ---------- Forwarded message ---------- > From: Prashanth Devarajappa <Prashanth.Devarajappa(a)enghouse.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Sep 2015 10:07:27 +0000 > Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold > > Thanks for your message. > > > > >> Best Route to do this is via session Manager as this is the supported > method. First create a trunk link between Session manager and Freeswitch > > > > This is how we have setup our environment and got this far. > > > > Prashanth > > > > > > *From:* kaleem rehman [mailto:k4kaleem@gmail.com] > *Sent:* 08 September 2015 10:46 > *To:* freeswitch-users(a)lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 111, Issue > 82 > > > > Hi Prashanth / Bob, > > > > Avaya CM supports sip but this is mainly used for Aura VM, Experience > portal and SM and more of Avaya only devices, direct conenctions to > Asterisk , FS & 3CX work but I always experienced issues due to Avaya SIP > Limitations and also officially this is not supported > > Best Route to do this is via session Manager as this is the supported > method. > > First create a trunk link between Session manager and Freeswitch, then > register extension directly to SM and make calls between from this > extension and Freeswitch, once everything is working then create a trunk > between CM & SM, or use existing and setup digit routing on SM and CM. > > > > if you have any issues setting up SM and its entities then let me know. > this way will be best and far most easy one. > > > > thanks, kaleem > > > > On Tue, Sep 8, 2015 at 9:54 AM, < > freeswitch-users-request(a)lists.freeswitch.org> wrote: > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: OneWay audio after hold->unhold (Bote Man) > 2. Re: Compiling under SmartOS (Stanislav Sinyagin) > 3. Re: OneWay audio after hold->unhold (Prashanth Devarajappa) > > > ---------- Forwarded message ---------- > From: Bote Man <bote_radio(a)botecomm.com> > To: "'FreeSWITCH Users Help'" <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 7 Sep 2015 18:03:17 -0400 > Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold > > Actually, yes I need all of the settings on the Avaya side. > > > > I am not using encryption and I never could get the Avaya CS1000M to talk > to the FreeSWITCH installation. I think the closest I could get was the > signaling would allow the call to be answered, but I got no audio to pass > either direction. > > > > I think the Avaya puts information in unexpected elements of the SIP > message and I had no more time to troubleshoot it because it was not my > primary task. > > > > There must be some small setting that I am missing and I would love to > find the missing piece of the puzzle. > > > > Thanks! > > > > Bote > > > > > > *From:* Prashanth Devarajappa > *Sent:* Monday, 07 September, 2015 09:47 > *Subject:* Re: [Freeswitch-users] OneWay audio after hold->unhold > > > > *Avaya PBX Version Control* > > § Media encryption over IP:Y > > § Session Manager Element Manager 6.3 > > § System Manager Release 6.3.5 > > § Communication System Management 6.3.10 > > > > *In IP-Network-region :* > > § Codec set matches group for encryption > > § Intra-region IP-IP Direct Audio: yes > > § Inter-region IP-IP Direct Audio: yes > > > > *In CM, signaling-group of type*: *SIP* > > § Transport Method: TLS > > > > Let me know if you need further Avaya config details. > > > > *FreeSwitch :* > > Its standard FS with custom (code/config) changes to enable > > > > · Support for RFC 5939 : Capability Negotiation > > · Support for UNENCRYPTED_SRTCP > > · Support IP Shuffling with in ACK…… to be done J > > > > > > Prashanth > > > > > > > > *From:* Bote Man [mailto:bote_radio@botecomm.com <bote_radio(a)botecomm.com>] > > *Sent:* 05 September 2015 04:50 > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] OneWay audio after hold->unhold > > > > We would be VERY appreciative if you could provide the configurations on > the Avaya and FreeSWITCH that allow them to talk to each other. In my > experience Avaya does not play well with others, and I have tried and given > up. > > Thank you in advance! > > Bote > > > > On Fri, Sep 4, 2015 at 12:09 PM, Prashanth Devarajappa < > Prashanth.Devarajappa(a)enghouse.com> wrote: > > Hello, > > > > I am connecting to Avaya switch using SIP trunk interface from FS with TLS > & SRTP connection. This works well for inbound and outbound calls. I have > an issue with media, when I try hold -> unhold operation from FS for an > outbound call. > > > > When FS tries to unhold the call by sending Re-INVITE, avaya responds with > 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to > make media flow directly b/w endpoints). FS responds with 200 OK + SDP as > expected. Now avaya sends ACK + SDP where SDP has the updated connection > info(IP and port of endpoint) where media should be sent. Along with this > it also changes the crypto master key as per RFC due to change in > connection info(c=). > > > > On seeing this ACK, FS update the connection info(which is good) and > change its own crypto master key in response. however this new master key > is not sent to far end, meaning FS can decode the media it receives but > other end can't decode the media FS is sending to it resulting in one way > audio. > > > > Has anyone come across this issue ? > > > > Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets > new master key ? > > > > > > Regards > > Prashanth > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > ---------- Forwarded message ---------- > From: Stanislav Sinyagin <ssinyagin(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Sep 2015 08:55:37 +0200 > Subject: Re: [Freeswitch-users] Compiling under SmartOS > See the update at https://freeswitch.org/jira/browse/FS-7967 > > I fixed the compilation problems, and now there's a runtime issue. > > On Mon, Aug 17, 2015 at 10:40 AM, Stanislav Sinyagin > <ssinyagin(a)gmail.com> wrote: > > I see there are some people on the list, working with SmartOS. > > > > The current master fails to compile: > > https://freeswitch.org/jira/browse/FS-7967 > > > > Your input will be appreciated. > > > > I just started looking around and getting the feeling what SmartOS is. > > I worked with Solaris quite a lot, but that was almost 10 years ago. > > > > > ---------- Forwarded message ---------- > From: Prashanth Devarajappa <Prashanth.Devarajappa(a)enghouse.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Sep 2015 08:53:42 +0000 > Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold > > Hi Bob, > > > > I am told that our Avaya is “avaya communication manager is the 8300” > > > > So far with our testing, we have not seen any issues related to transfer. > (May be we have not done extensive transfer tests) > > > > Prashanth > > > > > > *From:* Bob Hartwig [mailto:bobjectsfreeswitch@gmail.com] > *Sent:* 07 September 2015 17:42 > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] OneWay audio after hold->unhold > > > > Is this an Avaya AS5300 by any chance? Those have some serious interop > issues when it comes to transfers. I had to struggle mightily to get a > client's product (not Freeswitch) to interoperate with it. I'm really > interested to hear if this gets resolved for Freeswitch. > > > > > > > > On Fri, Sep 4, 2015 at 11:09 AM, Prashanth Devarajappa < > Prashanth.Devarajappa(a)enghouse.com> wrote: > > Hello, > > > > I am connecting to Avaya switch using SIP trunk interface from FS with TLS > & SRTP connection. This works well for inbound and outbound calls. I have > an issue with media, when I try hold -> unhold operation from FS for an > outbound call. > > > > When FS tries to unhold the call by sending Re-INVITE, avaya responds with > 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to > make media flow directly b/w endpoints). FS responds with 200 OK + SDP as > expected. Now avaya sends ACK + SDP where SDP has the updated connection > info(IP and port of endpoint) where media should be sent. Along with this > it also changes the crypto master key as per RFC due to change in > connection info(c=). > > > > On seeing this ACK, FS update the connection info(which is good) and > change its own crypto master key in response. however this new master key > is not sent to far end, meaning FS can decode the media it receives but > other end can't decode the media FS is sending to it resulting in one way > audio. > > > > Has anyone come across this issue ? > > > > Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets > new master key ? > > > > > > Regards > > Prashanth > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 111, Issue 82
by kaleem rehman 08 Sep '15

08 Sep '15

Hi Prashanth / Bob, Avaya CM supports sip but this is mainly used for Aura VM, Experience portal and SM and more of Avaya only devices, direct conenctions to Asterisk , FS & 3CX work but I always experienced issues due to Avaya SIP Limitations and also officially this is not supported Best Route to do this is via session Manager as this is the supported method. First create a trunk link between Session manager and Freeswitch, then register extension directly to SM and make calls between from this extension and Freeswitch, once everything is working then create a trunk between CM & SM, or use existing and setup digit routing on SM and CM. if you have any issues setting up SM and its entities then let me know. this way will be best and far most easy one. thanks, kaleem On Tue, Sep 8, 2015 at 9:54 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: OneWay audio after hold->unhold (Bote Man) > 2. Re: Compiling under SmartOS (Stanislav Sinyagin) > 3. Re: OneWay audio after hold->unhold (Prashanth Devarajappa) > > > ---------- Forwarded message ---------- > From: Bote Man <bote_radio(a)botecomm.com> > To: "'FreeSWITCH Users Help'" <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 7 Sep 2015 18:03:17 -0400 > Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold > > Actually, yes I need all of the settings on the Avaya side. > > > > I am not using encryption and I never could get the Avaya CS1000M to talk > to the FreeSWITCH installation. I think the closest I could get was the > signaling would allow the call to be answered, but I got no audio to pass > either direction. > > > > I think the Avaya puts information in unexpected elements of the SIP > message and I had no more time to troubleshoot it because it was not my > primary task. > > > > There must be some small setting that I am missing and I would love to > find the missing piece of the puzzle. > > > > Thanks! > > > > Bote > > > > > > *From:* Prashanth Devarajappa > *Sent:* Monday, 07 September, 2015 09:47 > *Subject:* Re: [Freeswitch-users] OneWay audio after hold->unhold > > > > *Avaya PBX Version Control* > > § Media encryption over IP:Y > > § Session Manager Element Manager 6.3 > > § System Manager Release 6.3.5 > > § Communication System Management 6.3.10 > > > > *In IP-Network-region :* > > § Codec set matches group for encryption > > § Intra-region IP-IP Direct Audio: yes > > § Inter-region IP-IP Direct Audio: yes > > > > *In CM, signaling-group of type*: *SIP* > > § Transport Method: TLS > > > > Let me know if you need further Avaya config details. > > > > *FreeSwitch :* > > Its standard FS with custom (code/config) changes to enable > > > > · Support for RFC 5939 : Capability Negotiation > > · Support for UNENCRYPTED_SRTCP > > · Support IP Shuffling with in ACK…… to be done J > > > > > > Prashanth > > > > > > > > *From:* Bote Man [mailto:bote_radio@botecomm.com <bote_radio(a)botecomm.com>] > > *Sent:* 05 September 2015 04:50 > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] OneWay audio after hold->unhold > > > > We would be VERY appreciative if you could provide the configurations on > the Avaya and FreeSWITCH that allow them to talk to each other. In my > experience Avaya does not play well with others, and I have tried and given > up. > > Thank you in advance! > > Bote > > > > On Fri, Sep 4, 2015 at 12:09 PM, Prashanth Devarajappa < > Prashanth.Devarajappa(a)enghouse.com> wrote: > > Hello, > > > > I am connecting to Avaya switch using SIP trunk interface from FS with TLS > & SRTP connection. This works well for inbound and outbound calls. I have > an issue with media, when I try hold -> unhold operation from FS for an > outbound call. > > > > When FS tries to unhold the call by sending Re-INVITE, avaya responds with > 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to > make media flow directly b/w endpoints). FS responds with 200 OK + SDP as > expected. Now avaya sends ACK + SDP where SDP has the updated connection > info(IP and port of endpoint) where media should be sent. Along with this > it also changes the crypto master key as per RFC due to change in > connection info(c=). > > > > On seeing this ACK, FS update the connection info(which is good) and > change its own crypto master key in response. however this new master key > is not sent to far end, meaning FS can decode the media it receives but > other end can't decode the media FS is sending to it resulting in one way > audio. > > > > Has anyone come across this issue ? > > > > Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets > new master key ? > > > > > > Regards > > Prashanth > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From: Stanislav Sinyagin <ssinyagin(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Sep 2015 08:55:37 +0200 > Subject: Re: [Freeswitch-users] Compiling under SmartOS > See the update at https://freeswitch.org/jira/browse/FS-7967 > > I fixed the compilation problems, and now there's a runtime issue. > > On Mon, Aug 17, 2015 at 10:40 AM, Stanislav Sinyagin > <ssinyagin(a)gmail.com> wrote: > > I see there are some people on the list, working with SmartOS. > > > > The current master fails to compile: > > https://freeswitch.org/jira/browse/FS-7967 > > > > Your input will be appreciated. > > > > I just started looking around and getting the feeling what SmartOS is. > > I worked with Solaris quite a lot, but that was almost 10 years ago. > > > > > ---------- Forwarded message ---------- > From: Prashanth Devarajappa <Prashanth.Devarajappa(a)enghouse.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 8 Sep 2015 08:53:42 +0000 > Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold > > Hi Bob, > > > > I am told that our Avaya is “avaya communication manager is the 8300” > > > > So far with our testing, we have not seen any issues related to transfer. > (May be we have not done extensive transfer tests) > > > > Prashanth > > > > > > *From:* Bob Hartwig [mailto:bobjectsfreeswitch@gmail.com] > *Sent:* 07 September 2015 17:42 > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] OneWay audio after hold->unhold > > > > Is this an Avaya AS5300 by any chance? Those have some serious interop > issues when it comes to transfers. I had to struggle mightily to get a > client's product (not Freeswitch) to interoperate with it. I'm really > interested to hear if this gets resolved for Freeswitch. > > > > > > > > On Fri, Sep 4, 2015 at 11:09 AM, Prashanth Devarajappa < > Prashanth.Devarajappa(a)enghouse.com> wrote: > > Hello, > > > > I am connecting to Avaya switch using SIP trunk interface from FS with TLS > & SRTP connection. This works well for inbound and outbound calls. I have > an issue with media, when I try hold -> unhold operation from FS for an > outbound call. > > > > When FS tries to unhold the call by sending Re-INVITE, avaya responds with > 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to > make media flow directly b/w endpoints). FS responds with 200 OK + SDP as > expected. Now avaya sends ACK + SDP where SDP has the updated connection > info(IP and port of endpoint) where media should be sent. Along with this > it also changes the crypto master key as per RFC due to change in > connection info(c=). > > > > On seeing this ACK, FS update the connection info(which is good) and > change its own crypto master key in response. however this new master key > is not sent to far end, meaning FS can decode the media it receives but > other end can't decode the media FS is sending to it resulting in one way > audio. > > > > Has anyone come across this issue ? > > > > Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets > new master key ? > > > > > > Regards > > Prashanth > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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