March 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_nibblebill question: DB Error while updating cash!
by Diego Viola 09 Mar '09

09 Mar '09

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: <variable name="nibble_rate" value="0.03"/> <variable name="nibble_account" value="18238"/> Any ideas? Thanks.

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Problem with a second incoming call to the same skype user name
by rdmitry 09 Mar '09

09 Mar '09

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… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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OpenZap and Sangoma A500 BRI card
by Sergey Kirillov 09 Mar '09

09 Mar '09

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[1000@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:1000@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 --- <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/>    </settings> <pri_spans> <span name="BRI_1">  <param name="q921loglevel" value="info"/> <param name="q931loglevel" value="info"/> <param name="mode" value="user"/> <param name="dialect" value="q931"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> </configuration> --- 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

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Please end the torment
by J Mann/Harry 09 Mar '09

09 Mar '09

No, I've yet to contribute anything, I barely have my system doing what I want. But I REALLY love Freeswitch and I want to see it BURY Asterisk. (Windows server user here) I've been struggling with the XML configs, trying to figure out what does what and where! That's fine, I'm used to it. What I'm NOT used to is the total lack of a forum-based community to join and participate in! Where can users SHARE their configs, help each other, learn from each others mistakes? No DEV forum? I'm speechless. Let's be serious guys, mailing lists went out in the 90s! IRC is "meh" at best! I want stickies, a forum for noobs, converts, a dev forum... So on... "WELCOME To FreeSWITCH!" Am I asking too much here? A FORUM? I can't see how you can spark interest when we're so sorely lacking the most basic and widely used community environments on the net! HELP SOMEBODY? Install SMF ASAP! http://www.simplemachines.org BTW: I hate people who voice problems without offering viable solutions in the process... Disgusting! So if someone can offer up a simple hosting account, Control Panel 10, Windows Linux whatever... I'll be more than happy to have SMF setup and receiving user registrations within 24 hours! I've done it dozens of times before. I will then gladly turn over the keys to the kingdom to the "powers that be" and take a backseat, being simply a happy user from that point on! Please folks, please. I'm dying over here and I'm sure I'm not alone! I'm searching Google and finding nothing!! FORUMS! Harry (email me here) switchserver(a)gmail.com (my FS email)

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Freeswitch IAX support
by Nik Middleton 09 Mar '09

09 Mar '09

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

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Codec problems about G7221 with Polycom IP550
by zhaoxxqq 09 Mar '09

09 Mar '09

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 9999. but with G7221 and G7221c have problems. I have config vars.xml to add <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@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

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Getting a sip trace on the console
by Nik Middleton 09 Mar '09

09 Mar '09

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,

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Contact header on INVITE
by Łukasz Czerpak 07 Mar '09

07 Mar '09

Hi, The INVITE request from my FreeSWITCH contains (1.2.3.4 - my public ip): Contact: <sip:gw+myprovider1@1.2.3.4:5080;transport=udp> but voip provider expects: Contact: <sip:username@1.2.3.4:5080;transport=udp> Is there any posiibility to change the Contact header value in gateway/dialplan configuration? Thanks, -- Łukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net]

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Prefered Linux Distro to run FS on
by Nik Middleton 07 Mar '09

07 Mar '09

Just curious here. I've always followed the fedora route but became disillusioned with the focus on the desktop rather than the server mode. Of late I've moved my servers to Centos. I felt the need for stable systems. Everyone seems to slate Centos, but to my surprise Anthony recommends Centos 5.2 which is nice to hear. Yes I know it's not bleeding edge, but I don't want that. Any reason why I should not be running Centos with FS? (I do plan on running 64 bit in future though) Regards,

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Lunchbox-type PC as small server?
by Fred 07 Mar '09

07 Mar '09

Hello I'm looking for a small, lunchbox-like PC to build a small-form factor CRM server to sell to small companies. Ideally, it should have one PCI slot so that I can stick a voice card to connect to an analog phone line and run FreeSwitch as well. I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it doesn't have room for a PCI slot, and I'm concerned about its performance. I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit pricey, and might also not be fast enough to act as a server. Are there brands/models you think I should look at? Thank you.

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