[Freeswitch-users] FS and Billing, AGI emulator
Hi, I wanted to use A2Billing on FS, but I noticed it uses some AGI stuff for dialling and to check how much credit the user has, etc. I heard you could use A2B by just importing the FS CDR data into it, but that wont work, so I come to the conclusion that I have no way of doing billing on FS yet. There is ASTPP but that's not complete yet, and I heard vague comments of doing an AGI emulator on top of the event socket on FS, how hard that would be? Is there a possibility to do that? How much would it cost? Thanks, Diego ___ 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] Script parsing a TPORT_DUMP sip log file to Mysql
Hi all,
I wrote a ruby script. it works for me. The script is in /scripts/
contrib/seven/sip/.
All ideas and suggestions are welcome!
Comment in script:
Now and then we need to look at sip traces to see want happened on a
failed call. There are lots of ways
to monitor sip messages. However, not all of them are convinient as
we want. Let's say a simple example:
FreeSWITCH :> originate sofia/gateways/carrier1/555|sofia/gateways/
carrier2/555|sofia/carrier3...
It's hard to tell what happend if the call fails. Because it's
different sip sessions.
The idea is to group them in one super session and see what happend. I
do this by
adding an arbitary sip header to do cross reference. And by parse the
sip messages to
a DB we can easily show it as html. I even can build a simple graph
based on the DB data:
http://skitch.com/seven1240/b8xj2/voip-master-idapted
You can easily add a sip header to INVITE by(I use x_interaction):
FreeSWITCH :> originate {sip_h_x_interaction=TEST0001}sofia/
gateways/.
And I can get all the messages from DB:
SELECT * FROM `sip_messages` WHERE (call_id IN
(SELECT distinct call_id FROM sip_messages WHERE
x_interaction = 'TEST0001')) ORDER BY created_at;
There are two aproches to get sip packets:
1) tcpdump/tshark
2) FreeSWITCH
I use the second. Note, there is no way to actually get sip messages
from FS currently, but sofia-sip
has the ability to log all sip messages to a disk file by using
TPORT_DUMP
And you can use this script to parse them to a DB. I know it hurt
performance,
but we don't have tons of traffic and you know there are only 5-10
messages for each
sip call. While we get about 1G bytes each day in the sip log, most of
them are OPTIONS/NOTIFY etc.
I filtered them before inserting to DB, but it will be better if sofia-
sip can filter that :)
The script will monitor the log file and parse and insert to DB in
real time. It's written in
the Ruby on Rails framework, however, I think it can run out of Rails
with or without
modification. But you still need ruby and rubygems if you want to use
it.
on Ubuntu/Debian
# apt-get install ruby rubygems
# gem install mysql file-tail yaml
It's just an idea, you may like to write your own tools to parse the
sip log file. Also
the log file need to be rotated regularly. And I think it maybe
possible to store the log
file on a memory disk, whatever... :)
Best
-Seven.
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Re: [Freeswitch-users] STUN error
Small joke :P Do you get that a lot? /b On Mar 9, 2009, at 10:09 PM, Will Smith wrote: Thank you Brian, it works like a champ. Yes, west philadelfia born and raised? ___ 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] STUN error
Thank you Brian, it works like a champ. Yes, west philadelfia born and raised? --- On Mon, 3/9/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] STUN error To: freeswitch-users@lists.freeswitch.org Date: Monday, March 9, 2009, 6:52 PM Sounds like DNS failure maybe... might wanna remove the ext-sip-ip and ext-rtp-ip setting out of external.xml to take care of that. west philadelfia born and raised? /b On Mar 9, 2009, at 8:36 PM, Will Smith wrote: > Hi, > I have the FS worked perfectly under NAT. And when I moved it to a > server with public IP, things getting wrong. > This is the error message that I got: > 2009-03-09 21:31:23 [ERR] sofia_glue.c:559 > sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: > 3478 [Remote Address Error!] > > -- ___ 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] STUN error
Will Smith wrote: > I tried whatever I can think of like; > set the > or > but still got the error. > Could you please give me some guide how to fix this. Change external_sip_ip and external_rtp_ip settings in vars.xml or in your external SIP profile. By default these are configured to use stun:stun.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] STUN error
Sounds like DNS failure maybe... might wanna remove the ext-sip-ip and ext-rtp-ip setting out of external.xml to take care of that. west philadelfia born and raised? /b On Mar 9, 2009, at 8:36 PM, Will Smith wrote: > Hi, > I have the FS worked perfectly under NAT. And when I moved it to a > server with public IP, things getting wrong. > This is the error message that I got: > 2009-03-09 21:31:23 [ERR] sofia_glue.c:559 > sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: > 3478 [Remote Address Error!] > > -- ___ 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] STUN error
Hi, I have the FS worked perfectly under NAT. And when I moved it to a server with public IP, things getting wrong. This is the error message that I got: 2009-03-09 21:31:23 [ERR] sofia_glue.c:559 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] -- I tried whatever I can think of like; set the or but still got the error. Could you please give me some guide how to fix this. 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
[Freeswitch-users] RPC and web admin panel for conference?
Hi all, I'm looking to implement an admin panel much like the one used at http://conference.freeswitch.org. Now I obviously cannot login and see the "admin" part of the panel but I'm pretty sure whats in there. I have xml_rpc running and can connect via http and issue commands. I've searched the forum here and went through the wiki, found nothing that looked like a panel. I was hoping to find a panel I can just configure and implement. Does anyone have a php (I guess, seeing as I have a php server) panel they can share with me? I'm sure I can get it working for my system. The thought of attempting one on my own at THIS point seems daunting at best. Any help would be greatly appreciated! 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
Re: [Freeswitch-users] mod_nibblebill question: DB Error while updating cash!
Ok, done. Thanks. On Mon, Mar 9, 2009 at 11:18 AM, Anthony Minessale wrote: > that means you should report it to jira not the mailing list. > > > On Sun, Mar 8, 2009 at 1:28 AM, Diego Viola wrote: >> >> Oh, I noticed the billing actually works, it discounts from my credit >> but I still get that message, even if the update works. >> >> "2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error >> while updating cash!" >> >> Thanks. >> >> On Sun, Mar 8, 2009 at 3:41 AM, Diego Viola wrote: >> > Hello, >> > >> > I'm trying to give mod_nibblebill a try, I compiled it and created the >> > DB, set up ODBC, etc. I'm using MySQL. >> > >> > This is how I created the db: >> > >> > CREATE TABLE accounts >> > ( >> > id int NOT NULL PRIMARY KEY, >> > name VARCHAR(255), >> > cash double precision NOT NULL >> > ); >> > >> > However when I try to make a call I get this: >> > >> > 2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error >> > while updating cash! >> > >> > I have this also on my user directory: >> > >> > >> > >> > >> > >> > Any ideas? >> > >> > 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 > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_miness...@hotmail.com > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@conference.freeswitch.org > 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 > > ___ 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] Problem with a second incoming call to the same skype user name
Hi Giovanni, Finally I was able to manage the problem. It was my fault. I didn't realized, that the value of the "name" parameter in this line: should strictly correspond to skype name you regester in startskype.sh script. It can not be arbitrary chosen, as I thought before. After I fixed that, everything works fine. I think you can put this point into skypiax page of freeswitch wiki. Best Regards, Dmitry Giovanni Maruzzelli-3 wrote: > > Thank you Dmitry, > > I'll have a look into it this evening (6 hours from now :-) ) > > > Sincerely, > > Giovanni Maruzzelli > = > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Mon, Mar 9, 2009 at 9:32 AM, rdmitry wrote: >> >> Hi Giovanni, >> >> I put everything you aked for in archive and attached it to the bug >> report >> at http://jira.freeswitch.org/browse/MODSKYPIAX-28 >> >> Hope it'll help to resolve this issue. >> >> Best Regards, Dmitry >> >> >> Giovanni Maruzzelli-3 wrote: >>> >>> Ciao Dmitry, >>> >>> The warnings are unharmful, I've just fixed them as per svn 12524, so >>> you will not see them anymore. But it will change nothing if there is >>> a problem (I mean, the warnings are not the problem and are not >>> indicating a problem). >>> >>> I cannot reproduce the problem, but maybe is because of the "strange >>> name problem". >>> >>> It would be of great help if you do, from the FS CLI: >>> >>> console loglevel 9 >>> >>> then reproduce the problem, and then attach (attach, not copy) *all* >>> the debug output (since beginning) to the Jira issue: >>> http://jira.freeswitch.org/browse/MODSKYPIAX-28 >>> >>> Ciao for now, >>> gm >>> >>> >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> = >>> www.celliax.org >>> via Pierlombardo 9, 20135 Milano >>> Italy >>> gmaruzz at celliax dot org >>> Cell : +39-347-2665618 >>> Fax : +39-02-87390039 >>> >>> >>> >>> >>> On Sun, Mar 8, 2009 at 11:17 AM, rdmitry wrote: Hi all, I've got a strange problem with skypiax. I successfully installed freeswitch revision 12408 with skypiax and configured 2 skype channels with different names. When I try to call both names one by one or simultaneously, everything goes fine. But when I try to place a second call to the same skype name which is busy with the first call, I get the following message: 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 ][skypiax1 ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized and second call can't get thru. I can hear call progress tones only. After about 5 seconds the message ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 ][skypiax1 ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized occurs and I can hear only silence after that. Does anybody know what might cause such a problem? I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) Any help would be very much appreciated. Best regards, Dmitry -- View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.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 >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22408941.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.fre
Re: [Freeswitch-users] OpenZap and Sangoma A500 BRI card
it's not released yet, please wait for the announcement that it has been completed sometime in the next week or 2. On Mon, Mar 9, 2009 at 1:41 PM, Sergey Kirillov wrote: > Hi everybody, > > I'm trying to use Sangoma A500 BRI card with OpenZap, but it does not work. > > Can somebody help to to configure it? > > Problem log (Incoming call): > > 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() > Unhandled event 2 > 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() > Unhandled event 2 > 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() > Unhandled event 2 > 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel OpenZAP/1:1/2360012 > [7473c92a-0a4e-11de-9dc3-c56d4d411902] > 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 80503820933->2360012 in context default > 2009-03-06 14:58:14 [NOTICE] switch_ivr.c:1343 > switch_ivr_session_transfer() Transfer OpenZAP/1:1/2360012 to > xml[1...@default] > 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 80503820933->1000 in context default > 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx > XML features > 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 > > record_session::/opt/freeswitch/recordings/80503820933.2009-03-06-14-58-14.wav > 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf > XML features > 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel > sofia/internal/sip:1...@192.168.122.1:5061;transport=udp > [748a2ba2-0a4e-11de-9dc3-c56d4d411902] > 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state > 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state > 2009-03-06 14:58:14 [NOTICE] switch_ivr_originate.c:1588 > switch_ivr_originate() Pre-Answer OpenZAP/1:1/2360012! > 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state > 2009-03-06 14:58:14 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received > unhandled message 125 (0x7d) > 2009-03-06 14:58:15 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received > unhandled message 125 (0x7d) > 2009-03-06 14:58:26 [NOTICE] Span:0 Q.921() I frame in invalid state > ignored > --- > > > Here are my config files > > --- openzap.conf -- > [span wanpipe BRI_1] > name => BRI_1 > trunk_type => bri > b-channel => 1:1-2 > d-channel => 1:3 > > > --- openzap.conf.xml --- > > > > > > > > > > > > > > > > > > > > > > --- wanpipe1.conf --- > [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > wp1aft1 = wanpipe1, auto, API, Comment > wp1aft2 = wanpipe1, auto, API, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 4 > PCIBUS= 5 > FE_MEDIA = E1 > FE_LCODE = HDB3 > FE_FRAME = CRC4 > FE_LINE = 1 > TE_CLOCK = NORMAL > TE_REF_CLOCK = 0 > TE_HIGHIMPEDANCE = NO > TE_RX_SLEVEL = 120 > LBO = 120OH > TE_SIG_MODE = CCS > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_HW_DTMF = NO > > [wp1aft1] > HDLC_STREAMING = NO > ACTIVE_CH = 1-15.17-31 > IDLE_FLAG = 0x7E > MTU = 240 > MRU = 240 > DATA_MUX= NO > TDMV_HWEC = NO > > [wp1aft2] > HDLC_STREAMING = YES > ACTIVE_CH = 16 > MTU = 1500 > MRU = 1500 > DATA_MUX= NO > TDMV_HWEC = NO > > > > ___ > 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:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org 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] OpenZap and Sangoma A500 BRI card
> [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > wp1aft1 = wanpipe1, auto, API, Comment > wp1aft2 = wanpipe1, auto, API, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 4 > PCIBUS = 5 > FE_MEDIA = E1 > FE_LCODE = HDB3 > FE_FRAME = CRC4 > FE_LINE = 1 > TE_CLOCK = NORMAL > TE_REF_CLOCK = 0 > TE_HIGHIMPEDANCE = NO > TE_RX_SLEVEL = 120 > LBO = 120OH > TE_SIG_MODE = CCS > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_HW_DTMF = NO > > [wp1aft1] > HDLC_STREAMING = NO > ACTIVE_CH = 1-15.17-31 > IDLE_FLAG = 0x7E > MTU = 240 > MRU = 240 > DATA_MUX = NO > TDMV_HWEC = NO > > [wp1aft2] > HDLC_STREAMING = YES > ACTIVE_CH = 16 > MTU = 1500 > MRU = 1500 > DATA_MUX = NO > TDMV_HWEC = NO > I'm no BRI expert but it looks to me like your wanpipe is set up for E1/EuroISDN. Where did you get this setup information? -MC ___ 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] OpenZap and Sangoma A500 BRI card
Hi everybody, I'm trying to use Sangoma A500 BRI card with OpenZap, but it does not work. Can somebody help to to configure it? Problem log (Incoming call): 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel OpenZAP/1:1/2360012 [7473c92a-0a4e-11de-9dc3-c56d4d411902] 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 80503820933->2360012 in context default 2009-03-06 14:58:14 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer OpenZAP/1:1/2360012 to xml[1...@default] 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 80503820933->1000 in context default 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/opt/freeswitch/recordings/80503820933.2009-03-06-14-58-14.wav 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/sip:1...@192.168.122.1:5061;transport=udp [748a2ba2-0a4e-11de-9dc3-c56d4d411902] 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [NOTICE] switch_ivr_originate.c:1588 switch_ivr_originate() Pre-Answer OpenZAP/1:1/2360012! 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2009-03-06 14:58:15 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2009-03-06 14:58:26 [NOTICE] Span:0 Q.921() I frame in invalid state ignored --- Here are my config files --- openzap.conf -- [span wanpipe BRI_1] name => BRI_1 trunk_type => bri b-channel => 1:1-2 d-channel => 1:3 --- openzap.conf.xml --- --- wanpipe1.conf --- [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] wp1aft1 = wanpipe1, auto, API, Comment wp1aft2 = wanpipe1, auto, API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS= 5 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 120 LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_HW_DTMF = NO [wp1aft1] HDLC_STREAMING = NO ACTIVE_CH = 1-15.17-31 IDLE_FLAG = 0x7E MTU = 240 MRU = 240 DATA_MUX= NO TDMV_HWEC = NO [wp1aft2] HDLC_STREAMING = YES ACTIVE_CH = 16 MTU = 1500 MRU = 1500 DATA_MUX= NO TDMV_HWEC = NO ___ 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] Please end the torment
Thank you, I appriciate that you get some benifits from our efforts. We only recommend irc because it's an easy-to-access multi user chat where we can put all of the people who need help in the same room in real time so they can help each other and we can help them. I tried not to get annoyed about the 2 cracks in previous posts from others about how old and outdated irc was. Everyone is entitled to their own cup of tea after all. But we don't have to retire protocols just because they are old? We still use SMTP and HTTP and FTP and don't mock them for their age. Conversely, I feel kinda the same way about the "web 2.0" farce where the same tired browser and js nightmares I faced in 1997 are now swept under the rug with a singe addition of a background http-get instead of actually re-inventing the wheel if you are going to bother calling it wheel 2.0 Yes you can do some cool new stuff, but not nearly as much as what you could have done in 10 years of effort towards a better way, too late now ;) But that's only my opinion i don't try to enstill it to anyone. On Sat, Mar 7, 2009 at 11:40 PM, Luis F Urrea wrote: > Same thing happened to me in regards IRC, I had not used it for years > before getting into FS, but as a total newbie I can say that IRC is really > good to get things going quickly and all those No rocket science questions > we have. I have had people helping realtime looking at pastebin's and > stuff, which is a really good thing. > > When one needs to elaborate on a question maybe mailing list/forum fits > better, but what I know in my experience is that eventually you will get the > answer thanks to the efforts of the community with the current set of tools. > > > ___ > 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:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org 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
[Freeswitch-users] Missing Diversion header in INVITE after 302 reply
Hello, following scenario: -Phone A is redirected unconditionally to phone C -Phone B calls A -Phone A replys with 302 and Dieversion header -FS detects the 302 and sends out a new INVITE to C I found that FS doesnt' include the received diversion sip header into the new INVITE sent to phone C. Is there a way to configure FS so that diversion header are included? Additionally: Is there a way to inform phone A about the diversion header, so that phone A get display a hint to user? regards Helmut ___ 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] Freeswitch IAX support
Hi, This issue is that our mod_iax is using the only freely available iax2 stack. A client library that was only designed for small softphones. There is a patch in jira to add registration support but it was not done correctly and we have not had much time to work on it. We've already had to add several unappealing hacks to the code we are using now to make it threadsafe and i don't think it will scale very far and you may find it a disappointment even with registration support. I had suggested at some point that we would consider making an entire new scalable implementation of iax2 designed as a client/server library but really it's probably the place of the authors of the protocol to provide such a resource. But if they do no wish to, I estimated the cost of developing such a stack to be in the range of 25k-35k. So the short answer is we have little to no demand for it, so we have not put much effort into supporting it. On Sun, Mar 8, 2009 at 7:11 AM, Nik Middleton < nik.middle...@noblesolutions.co.uk> wrote: > Hi Guys, > > > > Now that IAX has been published as an RFC ( > http://www.rfc-editor.org/authors/rfc5456.txt) are there any plans to > support registrations? > > > > Not a moan, just curious as to the road map. > > > > A lot of my users have Asterisk PBX’s using IAX and I’d love to replace my > Asterisk central server with FS to better serve them. Yes I know I could get > them to move to using SIP, but there’s a lot of them. > > > > Regards > > > > ___ > 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:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org 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] mod_nibblebill question: DB Error while updating cash!
that means you should report it to jira not the mailing list. On Sun, Mar 8, 2009 at 1:28 AM, Diego Viola wrote: > Oh, I noticed the billing actually works, it discounts from my credit > but I still get that message, even if the update works. > > "2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error > while updating cash!" > > Thanks. > > On Sun, Mar 8, 2009 at 3:41 AM, Diego Viola wrote: > > Hello, > > > > I'm trying to give mod_nibblebill a try, I compiled it and created the > > DB, set up ODBC, etc. I'm using MySQL. > > > > This is how I created the db: > > > > CREATE TABLE accounts > > ( > > id int NOT NULL PRIMARY KEY, > > name VARCHAR(255), > > cash double precision NOT NULL > > ); > > > > However when I try to make a call I get this: > > > > 2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error > > while updating cash! > > > > I have this also on my user directory: > > > > > > > > > > > > Any ideas? > > > > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org 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] Codec problems about G7221 with Polycom IP550
The IP550, 650 and 670 DO NOT support any G722.1 codecs at this point... expect support for those later in the year... right now they only support G722. /b On Mar 9, 2009, at 1:44 AM, zhaoxxqq wrote: Hello, I'm a newbe of Freeswitch. I have tried to config Polycom's soundpoint IP550 to use wideband codecs. G722 has no problem with conference and dial . but with G7221 and G7221c have problems. I have config vars.xml to add data="global_codec_prefs=g7...@32000h,g7...@16000h"/> and config polycom IP 550's SIP.cfg like the attachment. Can anyone help me to confirm if my config is right. Zhao Xiaoqiang 2009-03-09 zhaoxxqq ___ 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] Getting a sip trace on the console
Use SVN, or wait for the next release, fs_cli+siptrace rocks :) On Sun, Mar 8, 2009 at 12:03 PM, Nik Middleton wrote: > That's exactly what I was looking for, many thanks > > Regards, > > -Original Message- > From: freeswitch-users-boun...@lists.freeswitch.org > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of > Peter P GMX > Sent: 08 March 2009 12:58 > To: freeswitch-users@lists.freeswitch.org > Subject: Re: [Freeswitch-users] Getting a sip trace on the console > > I use the ngrep tool on the OS console and write the output to a file: > ngrep -d any port 5060 -W byline >outfile.txt > > Best regards > Peter > > Nik Middleton schrieb: >> >> Hi Guys, >> >> >> >> I'm trying to debug some SIP messaging issues. Is there a way of >> doing the Asterisk equivalent of SIP Debug so I can see what's being > sent? >> >> >> >> Regards, >> >> > >> >> ___ >> 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 > ___ 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] Problem with a second incoming call to the same skype user name
Thank you Dmitry, I'll have a look into it this evening (6 hours from now :-) ) Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Mon, Mar 9, 2009 at 9:32 AM, rdmitry wrote: > > Hi Giovanni, > > I put everything you aked for in archive and attached it to the bug report > at http://jira.freeswitch.org/browse/MODSKYPIAX-28 > > Hope it'll help to resolve this issue. > > Best Regards, Dmitry > > > Giovanni Maruzzelli-3 wrote: >> >> Ciao Dmitry, >> >> The warnings are unharmful, I've just fixed them as per svn 12524, so >> you will not see them anymore. But it will change nothing if there is >> a problem (I mean, the warnings are not the problem and are not >> indicating a problem). >> >> I cannot reproduce the problem, but maybe is because of the "strange >> name problem". >> >> It would be of great help if you do, from the FS CLI: >> >> console loglevel 9 >> >> then reproduce the problem, and then attach (attach, not copy) *all* >> the debug output (since beginning) to the Jira issue: >> http://jira.freeswitch.org/browse/MODSKYPIAX-28 >> >> Ciao for now, >> gm >> >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> = >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Sun, Mar 8, 2009 at 11:17 AM, rdmitry wrote: >>> >>> Hi all, >>> >>> I've got a strange problem with skypiax. I successfully installed >>> freeswitch >>> revision 12408 with skypiax and configured 2 skype channels with >>> different >>> names. When I try to call both names one by one or simultaneously, >>> everything goes fine. But when I try to place a second call to the same >>> skype name which is busy with the first call, I get the following >>> message: >>> >>> 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 >>> skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 >>> ][skypiax1 >>> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized >>> >>> and second call can't get thru. I can hear call progress tones only. >>> After >>> about 5 seconds the message >>> >>> ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 >>> skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 >>> ][skypiax1 >>> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized >>> >>> occurs and I can hear only silence after that. >>> >>> Does anybody know what might cause such a problem? >>> >>> I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) >>> >>> Any help would be very much appreciated. >>> >>> Best regards, Dmitry >>> >>> >>> >>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.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 >> >> > > -- > View this message in context: > http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22408941.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] Problem with a second incoming call to the same skype user name
Hi Giovanni, I put everything you aked for in archive and attached it to the bug report at http://jira.freeswitch.org/browse/MODSKYPIAX-28 Hope it'll help to resolve this issue. Best Regards, Dmitry Giovanni Maruzzelli-3 wrote: > > Ciao Dmitry, > > The warnings are unharmful, I've just fixed them as per svn 12524, so > you will not see them anymore. But it will change nothing if there is > a problem (I mean, the warnings are not the problem and are not > indicating a problem). > > I cannot reproduce the problem, but maybe is because of the "strange > name problem". > > It would be of great help if you do, from the FS CLI: > > console loglevel 9 > > then reproduce the problem, and then attach (attach, not copy) *all* > the debug output (since beginning) to the Jira issue: > http://jira.freeswitch.org/browse/MODSKYPIAX-28 > > Ciao for now, > gm > > > > Sincerely, > > Giovanni Maruzzelli > = > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Sun, Mar 8, 2009 at 11:17 AM, rdmitry wrote: >> >> Hi all, >> >> I've got a strange problem with skypiax. I successfully installed >> freeswitch >> revision 12408 with skypiax and configured 2 skype channels with >> different >> names. When I try to call both names one by one or simultaneously, >> everything goes fine. But when I try to place a second call to the same >> skype name which is busy with the first call, I get the following >> message: >> >> 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 >> skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 >> ][skypiax1 >> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized >> >> and second call can't get thru. I can hear call progress tones only. >> After >> about 5 seconds the message >> >> ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 >> skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 >> ][skypiax1 >> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized >> >> occurs and I can hear only silence after that. >> >> Does anybody know what might cause such a problem? >> >> I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) >> >> Any help would be very much appreciated. >> >> Best regards, Dmitry >> >> >> >> >> >> -- >> View this message in context: >> http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.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 > > -- View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22408941.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
