Re: [Freeswitch-users] How we can control calls using Java
The java api is 99% the same as the python/lua/perl api. You can check the wiki and the sample scripts in the source tree. This may be incomplete, feel free to ask any questions here where you can't find samples and we can try to fill in the missing pieces if they are not on the wiki. Mike On Aug 12, 2008, at 2:13 AM, Adeel Ansari wrote: Hi, I have managed to hook my Java program in. Now, looking for some hint, how we can control the call. Actually, I need to make a call, on genuine mobile phone, using Gizmo5 VoIP network. Furthermore, from where I can get the Mod-Java API documentation. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari ___ Freeswitch-users mailing list Freeswitch-users@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@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] Performance bottleneck
I’m trying to determine the FS resource bottleneck when operating under load (in windows environment), but can’t get the FS to load for some unseen reason. FS environment (a weak PC on purpose): CPU 2x Intel Pentium 4 3GHz RAM 2x 512MB DDR II RAM Chipset - Intel E7221 (Copper River) chipset ICH6R + FWH + BCM5721 LAN 1x Broadcom Giga LAN Windows 2003 Server – Service pack 2 FS version 9235 Running Release build on highest priority Load script: A different machine running sipP Running rtp_echo load, 50 cps, limit of 1000 calls, 30sec call duration, extension 9996 (echo test): sipp -rtp_echo -r 50 -l 1000 -d 3 -s 9996 -sf auc.xml -mp 25000 -i 192.168.1.1 -mi 192.168.1.1 192.168.1.2 Results: Test ran for 9.5 hours Total of 48828 calls - all successful No timeouts, retransmissions or unexpected messages. Peak was 1003 calls after 4563 seconds (actual 0.2 cps) Total of 1448750 RTP packets Average response time: 11min 21 seconds CPU usage 8% ~ 21%. Average 11%. Memory usage: Started with 26,000KB RAM, 27,660KB VM, 25 threads Peak at 136,000KB RAM,,367,004KB VM, 1024 threads Ended with 88,220KB RAM, 141,684KB VM, 24 threads Disk usage wasn’t monitored. My question is what is slowing the response time so much but keeps the CPU running low? NB Following Patrick Grondin’s post from 17-Jul-08, I intentionally didn’t change the default dialplan as I’m trying to load up the CPU. ___ Freeswitch-users mailing list Freeswitch-users@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] SDP issue receiving calls from SIP connection
Just played with the test_sdp tool ... looks like the issue might actually be the clock rate for format 125 ... tool says its invalid... Cheers Kirk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SDP issue receiving calls from SIP connection
To continue this... I've just noticed that all the rtpmap codec entries (from my Tesco INVITE) generally have a . in the token name ... it seems that sofia sdp_parse doesn't allow this. It seems valid according to the rfc 4566 ? Any comments ? Cheers Kirk 2008/8/12 Kirk Bateman [EMAIL PROTECTED] Just played with the test_sdp tool ... looks like the issue might actually be the clock rate for format 125 ... tool says its invalid... Cheers Kirk ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] GUI
try: http://www.wikipbx.org ;) Henk Oegema wrote: On Monday 04 August 2008 07:54:44 ram wrote: On Fri, Aug 1, 2008 at 8:02 AM, Lito Manansala [EMAIL PROTECTED]wrote: www.wikipbx.com Hi its asking Login, how can i download ram ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Can we have a forum
Well if you're reading this, this solution does appear to work. And is a win-win. I would recommend it. The interface on the web side is certainly friendly and easy-to-use. and then bkw doesn't have to actually do any work. Two thumbs up from me - Darren bsnipes wrote: I am coming in a little late on this conversation but you can have both with keeping the existing mailman system. I added freeswitch-users to Nabble groups back in March. The same thing can be done to freeswitch-dev and we might be able to import the even older freeswitch-users data. For those not familiar Nabble ( http://www.nabble.com ) allows anyone to create a web forum based on an existing mailing list. After you signup you can input your existing mailing list auth info and when you post to the freeswitch-users forum it sends it to the mailing list and keeps track of the posting status. Here is a link to the freeswitch list: http://www.nabble.com/Freeswitch-users-f32209.html This is a works for me solution that might appease everyone. Brian S/bsnipes On Sunday 10 August 2008 6:49:32 pm Ilan Perez wrote: I am in with the forum. For whatever reason people don't like them. Voip-info.org is working use it and people will start asking more questions on it and then it will be more popular etc.. My 2 cents Ilan Perez From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Holtsclaw Sent: 11 August 2008 07:36 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Can we have a forum Personally, I also prefer some sort of bulletin board / message board based discussion. Google Groups is a great idea. To me, a straight up mailman mailing list just seems old school. I agree with Darren in that something more easily searchable would be a big plus. On 8/10/2008 at 4:54 PM, Brian West [EMAIL PROTECTED] wrote: Oh btw I have talked about using google groups which offers both in one package What does everyone thing about some sort of compromise like that? /b On Aug 10, 2008, at 3:43 PM, Lee JJ wrote: Dear Sirs : Can we have a forum like phpbb or punbb . thanks ___ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/Can-we-have-a-forum-tp18916925p18944743.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Can we have a forum
Whoops, I guess it shows my handle instead of my name though... I'll have to go change that. -Original Message- From: RdFltErr [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 12, 2008 6:56 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Can we have a forum Well if you're reading this, this solution does appear to work. And is a win-win. I would recommend it. The interface on the web side is certainly friendly and easy-to-use. and then bkw doesn't have to actually do any work. Two thumbs up from me - Darren bsnipes wrote: I am coming in a little late on this conversation but you can have both with keeping the existing mailman system. I added freeswitch-users to Nabble groups back in March. The same thing can be done to freeswitch-dev and we might be able to import the even older freeswitch-users data. For those not familiar Nabble ( http://www.nabble.com ) allows anyone to create a web forum based on an existing mailing list. After you signup you can input your existing mailing list auth info and when you post to the freeswitch-users forum it sends it to the mailing list and keeps track of the posting status. Here is a link to the freeswitch list: http://www.nabble.com/Freeswitch-users-f32209.html This is a works for me solution that might appease everyone. Brian S/bsnipes On Sunday 10 August 2008 6:49:32 pm Ilan Perez wrote: I am in with the forum. For whatever reason people don't like them. Voip-info.org is working use it and people will start asking more questions on it and then it will be more popular etc.. My 2 cents Ilan Perez From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Holtsclaw Sent: 11 August 2008 07:36 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Can we have a forum Personally, I also prefer some sort of bulletin board / message board based discussion. Google Groups is a great idea. To me, a straight up mailman mailing list just seems old school. I agree with Darren in that something more easily searchable would be a big plus. On 8/10/2008 at 4:54 PM, Brian West [EMAIL PROTECTED] wrote: Oh btw I have talked about using google groups which offers both in one package What does everyone thing about some sort of compromise like that? /b On Aug 10, 2008, at 3:43 PM, Lee JJ wrote: Dear Sirs : Can we have a forum like phpbb or punbb . thanks ___ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org -- View this message in context: http://www.nabble.com/Can-we-have-a-forum-tp18916925p18944743.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Can we have a forum
They've started up Nabble 2 which apparently has newer interfaces available including phpbb. If you create an account you can add the freeswitch-users and freeswitch-dev to it and it should ask you to upload the archives. I can add freeswitch-users and tell them to transfer what is on Nabble v1 to it but since you have archives prior to March it would be better for you to do it. Brian S/bsnipes Brian West-3 wrote: Now lets figure out how to stuff the archives in to nabble :P /b On Aug 12, 2008, at 9:05 AM, bsnipes wrote: Replying back from Nabble :-) Darren Schreiber wrote: Well if you're reading this, this solution does appear to work. And is a win-win. I would recommend it. The interface on the web side is certainly friendly and easy-to-use. and then bkw doesn't have to actually do any work. Two thumbs up from me - Darren bsnipes wrote: I am coming in a little late on this conversation but you can have both with keeping the existing mailman system. I added freeswitch-users to Nabble groups back in March. The same thing can be done to freeswitch- dev and we might be able to import the even older freeswitch-users data. For those not familiar Nabble ( http://www.nabble.com ) allows anyone to create a web forum based on an existing mailing list. After you signup you can input your existing mailing list auth info and when you post to the freeswitch-users forum it sends it to the mailing list and keeps track of the posting status. Here is a link to the freeswitch list: http://www.nabble.com/Freeswitch-users-f32209.html This is a works for me solution that might appease everyone. Brian S/bsnipes On Sunday 10 August 2008 6:49:32 pm Ilan Perez wrote: I am in with the forum. For whatever reason people don't like them. Voip-info.org is working use it and people will start asking more questions on it and then it will be more popular etc.. My 2 cents Ilan Perez From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Holtsclaw Sent: 11 August 2008 07:36 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Can we have a forum Personally, I also prefer some sort of bulletin board / message board based discussion. Google Groups is a great idea. To me, a straight up mailman mailing list just seems old school. I agree with Darren in that something more easily searchable would be a big plus. On 8/10/2008 at 4:54 PM, Brian West [EMAIL PROTECTED] wrote: Oh btw I have talked about using google groups which offers both in one package What does everyone thing about some sort of compromise like that? /b On Aug 10, 2008, at 3:43 PM, Lee JJ wrote: Dear Sirs : Can we have a forum like phpbb or punbb . thanks ___ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/Can-we-have-a-forum-tp18916925p18944943.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/Can-we-have-a-forum-tp18916925p18945137.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] GUI
On Fri, Aug 1, 2008 at 9:51 AM, Anthony Minessale [EMAIL PROTECTED] wrote: ... A quad woodcrest 2.6ghz can do about 3000 simo media sessions with FS, the same box can just make it to 400 when they are all G729 transcoding calls. If they are bridged calls, that number goes in half, if we take media out of the picture that number quadruples. So I guess I could boast 400 CPS with 3000-6000 simo sessions, but what's the point, I'll let Ken do that.. ;) G729 transcoding? I thought there was no support for that... Anyway I can get it (other than writing it myself)? -- Nicolás Brenner ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [Sofia-sip-devel] SDP issue receiving calls from SIP connection
A partial solution, I've not figured everything out, but suspect it might cause issues elsewhere to do with codec selection, however ... I've added . to the sdp_parse (parse_rtpmap)... so in the token call it allows ALPHA DIGIT -. Works nicely now (so far), realised that all my codecs from the Tesco INVITE were just becoming G/bitrate Cheers Kirk 2008/8/12 Pekka Pessi [EMAIL PROTECTED] 2008/8/12 Kirk Bateman [EMAIL PROTECTED]: I've just noticed that all the rtpmap codec entries (from my Tesco INVITE) generally have a . in the token name ... it seems that sofia sdp_parse doesn't allow this. It seems valid according to the rfc 4566 ? The SDP parser has not been completely updated to RFC 4566. http://sofia-sip.sourceforge.net/refdocs/sofia_sip_conformance.html#4566 -- Pekka.Pessi mail at nokia.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Performance bottleneck
9996 is not a good test extension because it does not generate any audio unless it gets some. 9998 that generates a tone or make up an ext that plays a file is a better one. Processing of the sip calls can be delayed by the presence stuff which is very intensive, you can try turning it off and see if you get more calls. Also you should compare it to what happens with the test exten first in the dial plan. On Tue, Aug 12, 2008 at 2:58 AM, UV [EMAIL PROTECTED] wrote: I'm trying to determine the FS resource bottleneck when operating under load (in windows environment), but can't get the FS to load for some unseen reason. FS environment (a weak PC on purpose): CPU 2x Intel Pentium 4 3GHz RAM 2x 512MB DDR II RAM Chipset - Intel E7221 (Copper River) chipset ICH6R + FWH + BCM5721 LAN 1x Broadcom Giga LAN Windows 2003 Server – Service pack 2 FS version 9235 Running Release build on highest priority Load script: A different machine running sipP Running rtp_echo load, 50 cps, limit of 1000 calls, 30sec call duration, extension 9996 (echo test): sipp -rtp_echo -r 50 -l 1000 -d 3 -s 9996 -sf auc.xml -mp 25000 -i 192.168.1.1 -mi 192.168.1.1 192.168.1.2 Results: Test ran for 9.5 hours Total of 48828 calls - all successful No timeouts, retransmissions or unexpected messages. Peak was 1003 calls after 4563 seconds (actual 0.2 cps) Total of 1448750 RTP packets Average response time: 11min 21 seconds CPU usage 8% ~ 21%. Average 11%. Memory usage: Started with 26,000KB RAM, 27,660KB VM, 25 threads Peak at 136,000KB RAM,,367,004KB VM, 1024 threads Ended with 88,220KB RAM, 141,684KB VM, 24 threads Disk usage wasn't monitored. My question is what is slowing the response time so much but keeps the CPU running low? NB Following Patrick Grondin's post from 17-Jul-08, I intentionally didn't change the default dialplan as I'm trying to load up the CPU. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Performance bottleneck
Plus there are dozens of events fired per call leg to describe the presence changes which does not scale well to 50cps. This is one of the many shortcomings to SIP, the presence stuff is not scalable. On Tue, Aug 12, 2008 at 12:59 PM, Michael Jerris [EMAIL PROTECTED] wrote: It's going to be the disk io from sqlite. The presense states are all stored in sqlite (or odbc) data source. Mike On Aug 12, 2008, at 1:53 PM, UV wrote: Turning the presence off did the trick, although it would be important (to me, at least) to understand why as it changes the performance significantly. Is the presence mechanism waiting for some response from the network? I'm assuming it's waiting on something external because I couldn't find any CPU activity… -- *From:* [EMAIL PROTECTED] [ mailto:[EMAIL PROTECTED][EMAIL PROTECTED] ] *On Behalf Of *Anthony Minessale *Sent:* Wednesday, August 13, 2008 12:55 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Performance bottleneck 9996 is not a good test extension because it does not generate any audio unless it gets some. 9998 that generates a tone or make up an ext that plays a file is a better one. Processing of the sip calls can be delayed by the presence stuff which is very intensive, you can try turning it off and see if you get more calls. Also you should compare it to what happens with the test exten first in the dial plan. On Tue, Aug 12, 2008 at 2:58 AM, UV [EMAIL PROTECTED] wrote: I'm trying to determine the FS resource bottleneck when operating under load (in windows environment), but can't get the FS to load for some unseen reason. FS environment (a weak PC on purpose): CPU 2x Intel Pentium 4 3GHz RAM 2x 512MB DDR II RAM Chipset - Intel E7221 (Copper River) chipset ICH6R + FWH + BCM5721 LAN 1x Broadcom Giga LAN Windows 2003 Server – Service pack 2 FS version 9235 Running Release build on highest priority Load script: A different machine running sipP Running rtp_echo load, 50 cps, limit of 1000 calls, 30sec call duration, extension 9996 (echo test): sipp -rtp_echo -r 50 -l 1000 -d 3 -s 9996 -sf auc.xml -mp 25000 -i 192.168.1.1 -mi 192.168.1.1 192.168.1.2 Results: Test ran for 9.5 hours Total of 48828 calls - all successful No timeouts, retransmissions or unexpected messages. Peak was 1003 calls after 4563 seconds (actual 0.2 cps) Total of 1448750 RTP packets Average response time: 11min 21 seconds CPU usage 8% ~ 21%. Average 11%. Memory usage: Started with 26,000KB RAM, 27,660KB VM, 25 threads Peak at 136,000KB RAM,,367,004KB VM, 1024 threads Ended with 88,220KB RAM, 141,684KB VM, 24 threads Disk usage wasn't monitored. My question is what is slowing the response time so much but keeps the CPU running low? NB Following Patrick Grondin's post from 17-Jul-08, I intentionally didn't change the default dialplan as I'm trying to load up the CPU. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.6.1/1605 - Release Date: 11/08/2008 16:59 ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org
Re: [Freeswitch-users] Unable to receive calls from external sources.
Shortly after posting my original message it suddenly dawned on me
that my soft-client might be attempting to dial into my server using
port 5060 which i recalled was used, by default, for users in the
default context, not the public. Tacking :5080 onto the end of the
SIP URI i was dialing gave me a whole different set of errors. After
reading through the log messages and trying out some different
configuration i managed to get the basics working (accepting a call
from an external domain and passing it through to an internal user or
directing to voicemail if user isn't online).
I'll post my config here incase anyone else ever has this particular problem.
conf/directory/default/bert.xml:
include
user id=bert mailbox=2378
params
param name=password value=password/
param name=vm-password value=1000/
/params
variables
variable name=accountcode value=2378/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Bert/
variable name=effective_caller_id_number value=2378/
/variables
/user
/include
Added to conf/dialplan/public.xml:
extension name=bert
condition field=destination_number expression=^(bert|2378)$
action application=transfer data=bert XML default/
/condition
/extension
Create the file conf/dialplan/extensions/bert.xml:
include
extension name=bert
condition field=destination_number expression=^(bert|2378)$
action application=set data=dialed_ext=$1/
action application=export data=dialed_ext=$1/
/condition
condition field=destination_number expression=^${caller_id_number}$
action application=set
data=voicemail_authorized=${sip_authorized}/
action application=answer/
action application=sleep data=1000/
action application=voicemail data=check default $${domain}
${dialed_ext}/
!-- bind_meta_app can have these args key [a|b|ab]
[a|b|o|s] app --
anti-action application=bind_meta_app data=1 b s
execute_extension::dx XML features/
anti-action application=bind_meta_app data=2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav/
anti-action application=bind_meta_app data=3 b s
execute_extension::cf XML features/
anti-action application=set data=transfer_ringback=${uk-ring}/
anti-action application=set data=call_timeout=30/
!-- anti-action application=set
data=sip_exclude_contact=${network_addr}/ --
anti-action application=set data=hangup_after_bridge=true/
!--anti-action application=set
data=continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION/
--
anti-action application=set data=continue_on_fail=true/
anti-action application=db
data=insert/call_return/${dialed_ext}/${caller_id_number}/
anti-action application=db
data=insert/last_dial_ext/${dialed_ext}/${uuid}/
anti-action application=bridge data=user/[EMAIL PROTECTED]/
anti-action application=answer/
!--anti-action application=send_display data=Voicemail
for ${dialed_ext}/--
anti-action application=sleep data=1000/
anti-action application=voicemail data=default $${domain}
${dialed_ext}/
/condition
/extension
/include
This seems to work well, although the IVR before leaving a vm seems a
bit choppy (perhaps due to being on a Xen VPS?). The only question i
still have regarding this bit is whether it would be better to switch
the ports around so that FreeSWITCH accepts calls from external
domains on port 5060 and registrations within the default context on
port 5080 or do the majority of SIP servers look up and adhere to
what's specified in a domain's SRV records?
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Re: [Freeswitch-users] Performance bottleneck
That begs the question... is there a mechanism in sqlite or Linux that allows for the RAM drive to be backed up periodically? That would be a cool feature to get documented for those power users like Ken! ;) -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Rice Sent: Tuesday, August 12, 2008 11:07 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Performance bottleneck The Disk IO on sqlite can be quite a bit... One work around for this is to create a ram drive of sufficient size and mount it to /usr/local/freeswitch/db (or whatever your db dir is for freeswitch) this helps out greatly... But anything in the db will not be saved across system reboots unless you do something about that yourself K From: Michael Jerris [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Tue, 12 Aug 2008 13:59:13 -0400 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Performance bottleneck It's going to be the disk io from sqlite. The presense states are all stored in sqlite (or odbc) data source. Mike On Aug 12, 2008, at 1:53 PM, UV wrote: Turning the presence off did the trick, although it would be important (to me, at least) to understand why as it changes the performance significantly. Is the presence mechanism waiting for some response from the network? I'm assuming it's waiting on something external because I couldn't find any CPU activity... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] On Behalf Of Anthony Minessale Sent: Wednesday, August 13, 2008 12:55 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Performance bottleneck 9996 is not a good test extension because it does not generate any audio unless it gets some. 9998 that generates a tone or make up an ext that plays a file is a better one. Processing of the sip calls can be delayed by the presence stuff which is very intensive, you can try turning it off and see if you get more calls. Also you should compare it to what happens with the test exten first in the dial plan. On Tue, Aug 12, 2008 at 2:58 AM, UV [EMAIL PROTECTED] wrote: I'm trying to determine the FS resource bottleneck when operating under load (in windows environment), but can't get the FS to load for some unseen reason. FS environment (a weak PC on purpose): CPU 2x Intel Pentium 4 3GHz RAM 2x 512MB DDR II RAM Chipset - Intel E7221 (Copper River) chipset ICH6R + FWH + BCM5721 LAN 1x Broadcom Giga LAN Windows 2003 Server - Service pack 2 FS version 9235 Running Release build on highest priority Load script: A different machine running sipP Running rtp_echo load, 50 cps, limit of 1000 calls, 30sec call duration, extension 9996 (echo test): sipp -rtp_echo -r 50 -l 1000 -d 3 -s 9996 -sf auc.xml -mp 25000 -i 192.168.1.1 http://192.168.1.1 http://192.168.1.1 -mi 192.168.1.1 http://192.168.1.1 http://192.168.1.1 192.168.1.2 http://192.168.1.2 http://192.168.1.2 Results: Test ran for 9.5 hours Total of 48828 calls - all successful No timeouts, retransmissions or unexpected messages. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Performance bottleneck
Actually I don¹t know of any mechanism that will back up the DB... Where sqlite does work well for small to medium installations it only scales to a point... Sqlite does not reuse nodes¹ in the db on an update... It marks them as dead and creates a new entry... While this works ok on smaller tables w/ light to medium updates after a while you have to compress or vacuum the tables... This requires a table level lock with sqlite... FS does have some things built in to handle this, but under load this can cause the switch to appear to hang. Switching over to use something like Postgresql (my prefered db) helps out a good bit here, but keep in mind that in doing so you greatly increase the resources required for the db. Also don¹t forget that pgsql has a similar mechanism on how it handles updates, just don¹t forget to enable auto-vacuuming on pgsql... That is a discussion for a different list tho K From: Brian West [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Tue, 12 Aug 2008 13:24:40 -0500 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Performance bottleneck Well putting the db in ram does help a bit but it has to keep states of everything going on and do extra work for that... its a heavy task in itself. On Aug 12, 2008, at 1:19 PM, Michael Collins wrote: That begs the question is there a mechanism in sqlite or Linux that allows for the RAM drive to be backed up periodically? That would be a cool feature to get documented for those power users like Ken! ;) -MC Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] GUI
it's coming soon. On Tue, Aug 12, 2008 at 9:40 AM, Nicolas Brenner [EMAIL PROTECTED]wrote: On Fri, Aug 1, 2008 at 9:51 AM, Anthony Minessale [EMAIL PROTECTED] wrote: ... A quad woodcrest 2.6ghz can do about 3000 simo media sessions with FS, the same box can just make it to 400 when they are all G729 transcoding calls. If they are bridged calls, that number goes in half, if we take media out of the picture that number quadruples. So I guess I could boast 400 CPS with 3000-6000 simo sessions, but what's the point, I'll let Ken do that.. ;) G729 transcoding? I thought there was no support for that... Anyway I can get it (other than writing it myself)? -- Nicolás Brenner ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SDP issue receiving calls from SIP connection
Pekka Pessi from the sofia-sip project pushed a patch that is now synced to our tree. Please update and give it a try. Mike On Aug 11, 2008, at 11:41 AM, Michael Jerris wrote: Try posting this to the sofia-sip mailing list, lets see if we can sort out a fix there. Mike On Aug 11, 2008, at 11:25 AM, Kirk Bateman wrote: Michael, Great :( so probably something I'd need to mod the sofia source for really then. Not much chance of me getting Tesco (technically its a rebadge of freshtel / voicedot) to change their server so its compliant :) Cheers Kirk 2008/8/11 Michael Jerris [EMAIL PROTECTED] The issue from what I can see in the trace is the start of the s and o lines. We saw this before in a slightly different variant where those lines had extra whitespace in them after the =, this is probably the same thing, illegal chars after the =. Mike On Aug 11, 2008, at 11:09 AM, Kirk Bateman wrote: ldn't see anything specific that it was complaining about ... I've looked at the source and not figured it out yet... (really must try and memorize the spec someday). I was wondering if it was something to do with the X-NSE bit (dtmf tones extension to rfc ??) but given that I've set it to late negotiation I wouldn't expect the SDP parser to complain about that. I'm hoping the sofia dev can point me in the right direction. I have since I wrote the original mail managed to test (without any real changes) that I can make outgoing calls using the console originate command, that worked (no audio but I expected that). Cheers Kirk ___ Freeswitch-users mailing list Freeswitch-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Performance bottleneck
I dont know if this makes any sense - it's just an idea. If you're willing to take the hit of running MySQL, I know that it's replication features could potentially be used. You can have the primary MySQL server run in ramdisk and get all the performance benefits of doing so while also writing log files to the ram disk in a seperate area. Those logfiles can, using MySQL's built in replication features, be copied over to a backup server and played backup, giving you both a hot spare as well as a disk based backup. This does three things for you: 1) Gives you backup on disk, while preserving performance in RAM 2) Gives you a live backup that you can quickly shunt things over to if for some reason the primary dies 3) Allows you to handle spikes in volume. MySQL by default will just write to the log files and they can be played back later by the (slower) backup server, so a spike in volume of calls should not cause the server to slow down per say. There is a small risk your data will be lost if there is a failure for whatever is not copied over to the (slower) backup server, but that's unlikely to be that huge a lag (better then nothing). As to whether any of this applies (like why the heck you'd install MySQL on a ramdisk to start), I can't say. but it's a thought...Oh, and you need a lot of RAM ;-) _ From: Ken Rice [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 12, 2008 11:44 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Performance bottleneck Actually I don't know of any mechanism that will back up the DB... Where sqlite does work well for small to medium installations it only scales to a point... Sqlite does not reuse 'nodes' in the db on an update... It marks them as dead and creates a new entry... While this works ok on smaller tables w/ light to medium updates after a while you have to compress or vacuum the tables... This requires a table level lock with sqlite... FS does have some things built in to handle this, but under load this can cause the switch to appear to hang. Switching over to use something like Postgresql (my prefered db) helps out a good bit here, but keep in mind that in doing so you greatly increase the resources required for the db. Also don't forget that pgsql has a similar mechanism on how it handles updates, just don't forget to enable auto-vacuuming on pgsql... That is a discussion for a different list tho K _ From: Brian West [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Tue, 12 Aug 2008 13:24:40 -0500 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Performance bottleneck Well putting the db in ram does help a bit but it has to keep states of everything going on and do extra work for that... its a heavy task in itself. On Aug 12, 2008, at 1:19 PM, Michael Collins wrote: That begs the question. is there a mechanism in sqlite or Linux that allows for the RAM drive to be backed up periodically? That would be a cool feature to get documented for those power users like Ken! ;) -MC Brian West sip:[EMAIL PROTECTED] _ ___ Freeswitch-users mailing list Freeswitch-users@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@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] Failing to execute OS Command
Hi, I'm trying to convert recorded conversations from wav to mp3. I'm using ffmpeg for this purpose and i'm aiming to do this wthin Javascript. Yet, I'm encountering a problem. The conversion fails with the error message copied below. I must note that the same command works perfectly when executed from the command line. Any idea? Thanks, Klaus. ERROR MESSAGE== 2008-08-12 15:29:03 [NOTICE] mod_dptools.c:1169 system_session_function() Failed to execute command: ffmpeg -i /usr/local/freeswitch/recordings/test.wav -ab 32kb /usr/local/freeswitch/recordings/test.mp3 2008-08-12 15:29:03 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 16 (sofia/external/[EMAIL PROTECTED]) Ended 2008-08-12 15:29:03 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/external/[EMAIL PROTECTED] [CS_HANGUP] ==ERROR MESSAGE== -- Pt! Schon das coole Video vom GMX MultiMessenger gesehen? Der Eine für Alle: http://www.gmx.net/de/go/messenger03 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Failing to execute OS Command
Press F8 and try again so we see the entire debug log. /b On Aug 12, 2008, at 2:41 PM, Klaus Teller wrote: Hi, I'm trying to convert recorded conversations from wav to mp3. I'm using ffmpeg for this purpose and i'm aiming to do this wthin Javascript. Yet, I'm encountering a problem. The conversion fails with the error message copied below. I must note that the same command works perfectly when executed from the command line. Any idea? Thanks, Klaus. ERROR MESSAGE== 2008-08-12 15:29:03 [NOTICE] mod_dptools.c:1169 system_session_function() Failed to execute command: ffmpeg -i /usr/ local/freeswitch/recordings/test.wav -ab 32kb /usr/local/freeswitch/ recordings/test.mp3 2008-08-12 15:29:03 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 16 (sofia/external/[EMAIL PROTECTED] ) Ended 2008-08-12 15:29:03 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/external/[EMAIL PROTECTED] [CS_HANGUP] ==ERROR MESSAGE== -- Pt! Schon das coole Video vom GMX MultiMessenger gesehen? Der Eine für Alle: http://www.gmx.net/de/go/messenger03 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Failing to execute OS Command
Here is the command: session.execute(system, ffmpeg -i +file+.wav+ -ab 32kb +file+.mp3); I just tried with the full path and had the same result: session.execute(system, /usr/bin/ffmpeg -i +file+.wav+ -ab 32kb +file+.mp3); Klaus. -- GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen! Jetzt dabei sein: http://www.shortview.de/[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@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] Installation: good, but some issues
I've gotten FreeSWITCH running on this config:
- Intel Atom D945GCLF in 1U rackmount
- RedHat FC8
- Xorcom USB Astribank-8 (FXO)
- Xorcom USB Astribank-8 (FXS)
- Trunk FreeSWITCH code from a few days ago
- zaptel-1.4.9.2.xpp.r5566
- Grandstream BT-100 SIP phones
- Grandstream GXV-3000 SIP videophones
- Various analog phone extensions
First I tried installing on FC9. This didn't work because FC9 couldn't
deal with the on-board NIC card on the D945GCLF. FC8 installed fine,
though, so I just used that instead.
Overall I am very impressed with FreeSWITCH. It is vastly easier to set
up than Asterisk, and much cleaner. The XML config files are *so* much
better than the ad hoc Asterisk config syntax.
I like that I can make it run like a regular RedHat service, though I
could not get it to run as user freeswitch -- it seems to want to run as
root. I read through a bunch of the code and it is really well done.
Thanks for a great open source project!
Most things work: I set up the SIP phones and added the analog
extensions, and can dial out on any phone over my POTS lines via
Astribank FXO. I can receive incoming calls as well. I can
videoconference between the videophones by using the code in the sample
dialplan (default.xml) for intercom mode. Neat!
Problems:
Weirndess with FXS-originated calls
---
If I call from an analog (FXS) extension to a BT-100, it rings normally.
But if I then pick up the SIP phone, I can hear the ring on the analog
extension get choppy. Soon after, the call is dropped.
If I call from an analog (FXS) extension to a GXV-3000, it rings
normally, but when I pick up the SIP phone, I get an immediate busy on
the analog extension.
If I call from an analog (FXS) extension to an X-Lite SoftPhone, it
rings normally, but when I pick up the SIP phone, no audio comes through
from the analog extension.
If I call from either a BT-100, a GXV-3000, or an X-Lite SoftPhone *to*
an analog (FXS) extension, it works fine.
The dialplan is simple (note that I have 1- and 2-digit extensions):
!-- dial an OpenZAP
channel number to get the corresponding analog extension --
extension name=OpenZAP extensions
condition field=destination_number expression=^(9|1[0-6])$
action application=set data=dialed_ext=$1/
action application=bridge data=OpenZAP/${dialed_ext}/1/
/condition
/extension
!-- dial a number in the directory to get the corresponding SIP
extension; use video if possible --
extension name=local-extension
condition field=destination_number expression=^([3-6]\d)$
action application=set data=dialed_ext=$1/
action
application=export![CDATA[sip_h_Call-Info=sip:$${domain};answer-after=0]]/action
action application=export
data=sip_invite_params=intercom=true/
action application=export data=sip_auto_answer=true/
action application=bridge
data=user/[EMAIL PROTECTED]/
/condition
/extension
In general, I found it a bit strange that you can't (it seems) put
analog extensions into the directory. Is it true that you have to define
analog extensions manually in the dialplan? That's kind of
counterintuitive, especially for Asterisk users; the Asterisk Now GUI
makes analog and SIP extensions look the same for all practical purposes.
Echo
I was getting massive distortion on analog connections involving any
GXV-3000 before I changed echo_cancel_level in /etc/openzap/zt.conf. The
default value was 64. I changed it to 0 and the distortion went away,
but of course I get lots of annoying echo now. Other SIP phones don't
seem to have this problem. I have read in earlier posts to this list
that the GXV-3000 has known problems. But the interesting thing is that
it was working fine with Asterisk and the Astribanks (though on a
different CPU M/B).
The Astribank seems to have its own EC, but I can't figure out how to
turn it on. And I have no idea what to set the value in zt.conf to; 32
seems to work better than either 16 or 64 (!).
I tried fxotune but that didn't do anything: it ran for a long time and
generated an /etc/fxotone that seemed to have all zero values.
Basically, it would be great to have an echo cancellation HOWTO for
OpenZap users since EC seems to be offered in several different layers.
I also notice a bunch of different EC modes in the OpenZap code -- does
anyone know how to try different ones out?
FAX detection
-
This just doesn't seem to work. Here's the relevant code from my
diaplan. (It looks a little odd because I send all unanswered calls from
my POTS lines to the ext. 31 voicemail box.)
!-- ring all extensions for 20 seconds, then send to voicemail for
extension 31 --
!-- we answer immediately to prevent the alarm from emitting a
fax/modem tone in
Re: [Freeswitch-users] Installation: good, but some issues
I would recommend you open a Jira on the audio issue.
http://jira.freeswitch.org
so we don't loose track of the issue. I know there is a problem
with the astribank not hanging up properly.
/b
On Aug 12, 2008, at 5:31 PM, David Baggett wrote:
I've gotten FreeSWITCH running on this config:
- Intel Atom D945GCLF in 1U rackmount
- RedHat FC8
- Xorcom USB Astribank-8 (FXO)
- Xorcom USB Astribank-8 (FXS)
- Trunk FreeSWITCH code from a few days ago
- zaptel-1.4.9.2.xpp.r5566
- Grandstream BT-100 SIP phones
- Grandstream GXV-3000 SIP videophones
- Various analog phone extensions
First I tried installing on FC9. This didn't work because FC9 couldn't
deal with the on-board NIC card on the D945GCLF. FC8 installed fine,
though, so I just used that instead.
Overall I am very impressed with FreeSWITCH. It is vastly easier to
set
up than Asterisk, and much cleaner. The XML config files are *so* much
better than the ad hoc Asterisk config syntax.
I like that I can make it run like a regular RedHat service, though I
could not get it to run as user freeswitch -- it seems to want to
run as
root. I read through a bunch of the code and it is really well done.
Thanks for a great open source project!
Most things work: I set up the SIP phones and added the analog
extensions, and can dial out on any phone over my POTS lines via
Astribank FXO. I can receive incoming calls as well. I can
videoconference between the videophones by using the code in the
sample
dialplan (default.xml) for intercom mode. Neat!
Problems:
Weirndess with FXS-originated calls
---
If I call from an analog (FXS) extension to a BT-100, it rings
normally.
But if I then pick up the SIP phone, I can hear the ring on the analog
extension get choppy. Soon after, the call is dropped.
If I call from an analog (FXS) extension to a GXV-3000, it rings
normally, but when I pick up the SIP phone, I get an immediate busy on
the analog extension.
If I call from an analog (FXS) extension to an X-Lite SoftPhone, it
rings normally, but when I pick up the SIP phone, no audio comes
through
from the analog extension.
If I call from either a BT-100, a GXV-3000, or an X-Lite SoftPhone
*to*
an analog (FXS) extension, it works fine.
The dialplan is simple (note that I have 1- and 2-digit extensions):
!-- dial an OpenZAP
channel number to get the corresponding analog extension --
extension name=OpenZAP extensions
condition field=destination_number expression=^(9|1[0-6])$
action application=set data=dialed_ext=$1/
action application=bridge data=OpenZAP/${dialed_ext}/1/
/condition
/extension
!-- dial a number in the directory to get the corresponding SIP
extension; use video if possible --
extension name=local-extension
condition field=destination_number expression=^([3-6]\d)$
action application=set data=dialed_ext=$1/
action
application=export![CDATA[sip_h_Call-Info=sip:$$
{domain};answer-after=0]]/action
action application=export
data=sip_invite_params=intercom=true/
action application=export data=sip_auto_answer=true/
action application=bridge
data=user/[EMAIL PROTECTED]/
/condition
/extension
In general, I found it a bit strange that you can't (it seems) put
analog extensions into the directory. Is it true that you have to
define
analog extensions manually in the dialplan? That's kind of
counterintuitive, especially for Asterisk users; the Asterisk Now GUI
makes analog and SIP extensions look the same for all practical
purposes.
Echo
I was getting massive distortion on analog connections involving any
GXV-3000 before I changed echo_cancel_level in /etc/openzap/zt.conf.
The
default value was 64. I changed it to 0 and the distortion went away,
but of course I get lots of annoying echo now. Other SIP phones don't
seem to have this problem. I have read in earlier posts to this list
that the GXV-3000 has known problems. But the interesting thing is
that
it was working fine with Asterisk and the Astribanks (though on a
different CPU M/B).
The Astribank seems to have its own EC, but I can't figure out how to
turn it on. And I have no idea what to set the value in zt.conf to; 32
seems to work better than either 16 or 64 (!).
I tried fxotune but that didn't do anything: it ran for a long time
and
generated an /etc/fxotone that seemed to have all zero values.
Basically, it would be great to have an echo cancellation HOWTO for
OpenZap users since EC seems to be offered in several different
layers.
I also notice a bunch of different EC modes in the OpenZap code --
does
anyone know how to try different ones out?
FAX detection
-
This just doesn't seem to work. Here's the relevant code from my
diaplan. (It looks a
Re: [Freeswitch-users] Installation: good, but some issues
I had no problem with FXS to x-lite, do you have the latest SVN trunk or one
of the tarballs? We may have fixed some issues if you have an older release.
We have an application you can use in the dialplan called set_user
for example put this as the very first extension in the dialplan
extension name=set_openzap_user continue=true
condition field=source expression=mod_openzap/
action application=set_user data=[EMAIL PROTECTED]/
/condition
/extension
if you put that in an extension at the top of your dialplan just for calls
from openzap then you
will make that call assume the settings of the user in the directory with an
id that matches the caller id num set in openzap then fall through to the
rest of the dialplan.
most things you *think* are not possible are just not forced into place =D
as for astribank, I have heard there are a few issues with them and I do not
have one to test them. The guy who supports them is around on IRC but he
tends to favor the AST that's why he calls them ASTribank so we can try to
work with him if he's willing.
On Tue, Aug 12, 2008 at 5:31 PM, David Baggett [EMAIL PROTECTED]wrote:
I've gotten FreeSWITCH running on this config:
- Intel Atom D945GCLF in 1U rackmount
- RedHat FC8
- Xorcom USB Astribank-8 (FXO)
- Xorcom USB Astribank-8 (FXS)
- Trunk FreeSWITCH code from a few days ago
- zaptel-1.4.9.2.xpp.r5566
- Grandstream BT-100 SIP phones
- Grandstream GXV-3000 SIP videophones
- Various analog phone extensions
First I tried installing on FC9. This didn't work because FC9 couldn't
deal with the on-board NIC card on the D945GCLF. FC8 installed fine,
though, so I just used that instead.
Overall I am very impressed with FreeSWITCH. It is vastly easier to set
up than Asterisk, and much cleaner. The XML config files are *so* much
better than the ad hoc Asterisk config syntax.
I like that I can make it run like a regular RedHat service, though I
could not get it to run as user freeswitch -- it seems to want to run as
root. I read through a bunch of the code and it is really well done.
Thanks for a great open source project!
Most things work: I set up the SIP phones and added the analog
extensions, and can dial out on any phone over my POTS lines via
Astribank FXO. I can receive incoming calls as well. I can
videoconference between the videophones by using the code in the sample
dialplan (default.xml) for intercom mode. Neat!
Problems:
Weirndess with FXS-originated calls
---
If I call from an analog (FXS) extension to a BT-100, it rings normally.
But if I then pick up the SIP phone, I can hear the ring on the analog
extension get choppy. Soon after, the call is dropped.
If I call from an analog (FXS) extension to a GXV-3000, it rings
normally, but when I pick up the SIP phone, I get an immediate busy on
the analog extension.
If I call from an analog (FXS) extension to an X-Lite SoftPhone, it
rings normally, but when I pick up the SIP phone, no audio comes through
from the analog extension.
If I call from either a BT-100, a GXV-3000, or an X-Lite SoftPhone *to*
an analog (FXS) extension, it works fine.
The dialplan is simple (note that I have 1- and 2-digit extensions):
!-- dial an OpenZAP
channel number to get the corresponding analog extension --
extension name=OpenZAP extensions
condition field=destination_number expression=^(9|1[0-6])$
action application=set data=dialed_ext=$1/
action application=bridge data=OpenZAP/${dialed_ext}/1/
/condition
/extension
!-- dial a number in the directory to get the corresponding SIP
extension; use video if possible --
extension name=local-extension
condition field=destination_number expression=^([3-6]\d)$
action application=set data=dialed_ext=$1/
action
application=export![CDATA[sip_h_Call-Info=sip:$${domain};answer-after=0]]/action
action application=export
data=sip_invite_params=intercom=true/
action application=export data=sip_auto_answer=true/
action application=bridge
data=user/[EMAIL PROTECTED]/
/condition
/extension
In general, I found it a bit strange that you can't (it seems) put
analog extensions into the directory. Is it true that you have to define
analog extensions manually in the dialplan? That's kind of
counterintuitive, especially for Asterisk users; the Asterisk Now GUI
makes analog and SIP extensions look the same for all practical purposes.
Echo
I was getting massive distortion on analog connections involving any
GXV-3000 before I changed echo_cancel_level in /etc/openzap/zt.conf. The
default value was 64. I changed it to 0 and the distortion went away,
but of course I get lots of annoying echo now. Other SIP phones don't
seem to have this problem. I have read in earlier posts to this list
that the GXV-3000
Re: [Freeswitch-users] Installation: good, but some issues
Ask Anthony how much time he and the other devs actually *thought*
about what they were going to do *before* they started hacking code!
The FS devs did more than just learn from Asterisk - they made
practical application of those lessons.
-MC
Sent from my iPhone
On Aug 12, 2008, at 6:00 PM, David Baggett [EMAIL PROTECTED]
wrote:
I used an SVN checkout from a few days ago -- Friday, I think.
Thanks for the tip on the set_openzap_user app -- that's very cool.
Despite my comment about the directory, In general I'm actually
impressed with your foresight -- it's clear you've done a lot of the
right things in abstracting away from the original Asterisk model.
Should be a great platform for future ideas.
Dave
Anthony Minessale wrote:
I had no problem with FXS to x-lite, do you have the latest SVN
trunk or
one of the tarballs? We may have fixed some issues if you have an
older
release.
We have an application you can use in the dialplan called set_user
for example put this as the very first extension in the dialplan
extension name=set_openzap_user continue=true
condition field=source expression=mod_openzap/
action application=set_user data=[EMAIL PROTECTED]
{domain}/
/condition
/extension
if you put that in an extension at the top of your dialplan just for
calls from openzap then you
will make that call assume the settings of the user in the directory
with an id that matches the caller id num set in openzap then fall
through to the rest of the dialplan.
most things you *think* are not possible are just not forced into
place =D
as for astribank, I have heard there are a few issues with them and
I do
not have one to test them. The guy who supports them is around on
IRC
but he tends to favor the AST that's why he calls them ASTribank so
we
can try to work with him if he's willing.
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Re: [Freeswitch-users] How we can control calls using Java
Presently, I am trying to make a call using Java. Now, a little of my background. I have never used Asterisk before. Just used Brekeke PBX and SIP Server. That goes as, develop a webservice client and initiate the call. The method/function accepts argument user-agent, caller/sender (Registered on SIPServer), callee/recepient array(Hardphones). Then, caller would call first callee and then the second and then bridge the session. It was quite straight. Here I am finding it difficult to understand, I mean where to start from. Actually, I don't know what class, which methods are exposed for XML-RPC or through any web service. Right now, I am using Java Program as hook. I am making a call using Twinkle(SIP Client). Then in my program I am trying to originate a new session and then bridge. But I am getting * [ERR] mod_sofia.c:1946 sofia_outgoing_channel() Invalid Profile* on *session.originate(newSession1, sofia/sip/ [EMAIL PROTECTED]);* Any hint. Please guide me. Am I going in the right direction? Whats missing? Where to define phone no.? In my case its in the sip URL. Thanks. On Tue, Aug 12, 2008 at 2:28 PM, Michael Jerris [EMAIL PROTECTED] wrote: The java api is 99% the same as the python/lua/perl api. You can check the wiki and the sample scripts in the source tree. This may be incomplete, feel free to ask any questions here where you can't find samples and we can try to fill in the missing pieces if they are not on the wiki. Mike On Aug 12, 2008, at 2:13 AM, Adeel Ansari wrote: Hi, I have managed to hook my Java program in. Now, looking for some hint, how we can control the call. Actually, I need to make a call, on genuine mobile phone, using Gizmo5 VoIP network. Furthermore, from where I can get the Mod-Java API documentation. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari ___ Freeswitch-users mailing list Freeswitch-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
