November 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FXS doesn't stop ringing
by Komar, Jason 28 Nov '11

28 Nov '11

Hello Everyone, My current setup uses a Sangoma A400 card with 2 FXO and 2 FXS ports. I am working toward having my analog lines ported over to SIP trunks, but in the meantime I am working with what I have. When a call comes in over the POTS, my public dialplan transfers the call to extension 2000 which is setup in my default dialplan. This extension is setup like the Local_Extension example, but instead of bridging with this line: <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> I bridge to my FXS port: <action application="bridge" data="freetdm/1/1"/> It seems to work well, but the issue I have is that the FXS port keeps ringing the analog phone even after the call bridges to voicemail. I also tested doing a call in and hangup after 1 ring and the FXS keeps ringing and does not detect the hangup. I appreciate any help that someone can offer. Thanks, Jason Komar

2 1

Help Making a Call through External Sip Provider
by daniel mican 28 Nov '11

28 Nov '11

Greetings, I am having difficulty making external phone calls through freeswitch. I successfully receive calls but cannot sned them. I believe the problem is somewhere in here, I am not resolving contact with my sip gateway. It just keeps trying to invite until it times out. I have double checked all config files, and spent time reading the fs wiki's in hope of find what I am doing wrong. Below is the log and a packet. I have been fumbling with this for days and any help at all would be greatly appreciated. Ty 2011-11-24 09:35:13.274794 [DEBUG] sofia.c:5366 Channel sofia/external/+14102944410 entering state [calling][0] nua: nua_handle_magic: entering nta: timer A fired, retransmit INVITE (20736592) tport_release(0x7f7b2c052c00): 0x7f7b2c10b940 by 0x7f7b2c074b10 with (nil) tport_tsend(0x7f7b2c052c00) tpn = UDP/75.101.162.138:5060 tport_resolve addrinfo = 75.101.162.138:5060 tport_by_addrinfo(0x7f7b2c052c00): not found by name UDP/75.101.162.138:5060 tport_vsend(0x7f7b2c052c00): 1089 bytes of 1089 to udp/75.101.162.138:5060 tport_vsend returned 1089 nta: resent INVITE (20736592) to UDP/75.101.162.138:5060 tport_pend(0x7f7b2c052c00): pending 0x7f7b2c10b940 for udp/10.0.0.4:5080(already 0) nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/1 term, 0/2 free nta: timer set next to 2000 ms nta: timer A fired, retransmit INVITE (20736592) tport_release(0x7f7b2c052c00): 0x7f7b2c10b940 by 0x7f7b2c074b10 with (nil) tport_tsend(0x7f7b2c052c00) tpn = UDP/75.101.162.138:5060 tport_resolve addrinfo = 75.101.162.138:5060 tport_by_addrinfo(0x7f7b2c052c00): not found by name UDP/75.101.162.138:5060 tport_vsend(0x7f7b2c052c00): 1089 bytes of 1089 to udp/75.101.162.138:5060 tport_vsend returned 1089 nta: resent INVITE (20736592) to UDP/75.101.162.138:5060 tport_pend(0x7f7b2c052c00): pending 0x7f7b2c10b940 for udp/10.0.0.4:5080(already 0) nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/1 term, 0/2 free nta: timer set next to 4000 ms nta: timer A fired, retransmit INVITE (20736592) tport_release(0x7f7b2c052c00): 0x7f7b2c10b940 by 0x7f7b2c074b10 with (nil) tport_tsend(0x7f7b2c052c00) tpn = UDP/75.101.162.138:5060 tport_resolve addrinfo = 75.101.162.138:5060 tport_by_addrinfo(0x7f7b2c052c00): not found by name UDP/75.101.162.138:5060 tport_vsend(0x7f7b2c052c00): 1089 bytes of 1089 to udp/75.101.162.138:5060 tport_vsend returned 1089 nta: resent INVITE (20736592) to UDP/75.101.162.138:5060 tport_pend(0x7f7b2c052c00): pending 0x7f7b2c10b940 for udp/10.0.0.4:5080(already 0) 09:35:28.281385 IP (tos 0x0, ttl 64, id 12974, offset 0, flags [none], proto UDP (17), length 1117) 10.0.0.4.5080 > 75.101.162.138.5060: [udp sum ok] UDP, length 1089 E..]2...@.K. ...Ke.......I..INVITE sip:+14102944410@sip1.teltechsys.com SIP/2.0 Via: SIP/2.0/UDP 68.48.6.208:5080;rport;branch=z9hG4bKZ2Q3Km0KNKUcK Max-Forwards: 70 From: "" <sip:+14102944410@68.48.6.208>;tag=3pmg8NZt4SHNQ To: <sip:+14102944410@sip1.teltechsys.com> Call-ID: 5c4496eb-914c-122f-9d94-0025646e9c00 CSeq: 20736592 INVITE Contact: <sip:gw+sip1@68.48.6.208:5080;transport=udp;gw=sip1> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-68627e8 2011-11-21 13-52-28 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-Carrier: sip.flowroute.com X-Carrier-Prefix: 74278017 X-FS-Support: update_display Remote-Party-ID: <sip:+14102944410@68.48.6.208 >;party=calling;screen=yes;privacy=off

2 1

Soft phones could not register to my Freeswitch instance
by ayobami 28 Nov '11

28 Nov '11

I have my Freeswitch box connected to some other computer through wireless, FS automatically detects the IP address of the wireless adapter and things work fine every soft phone could register and calls were routed appropriately. But when I connected the Freeswitch box to a switch with the other computers many issues came up. At first freeswitch does not bind to the ethernet adapter, it indicates that its binding to the loopback IP address 127.0.0.1, so I tried restarting several times, no luck, so I decided to do some tweaking in Vars.xml I changed this <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> to <X-PRE-PROCESS cmd="set" data="domain=10.0.0.10"/> and in the internal.xml in sip_profile folder I changed these two settings from default values to so after all these changes, FS now binds to the ethernet adapter. But the problem now is no softphone on other computers could register again, they are all giving error 408 which means they could not register, I have googled and looked for solutions, but couldnot find any, please people help me out, I am really frustrated -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Soft-phones-could-not-registe… Sent from the freeswitch-users mailing list archive at Nabble.com.

3 2

Re: [Freeswitch-users] FreeTDM does not work
by Valery Kalinin 28 Nov '11

28 Nov '11

> Just out of curiosity, did you do a reboot and get a kernel change? I > had this happen to me and it was because only part of the dahdi stuff > (which I recompile on every reboot to catch this now) got compiled and > things broke that way. I did not understand a bit. You offer to recompile dahdi?

2 1

Soft phones could not register to my Freeswitch instance
by ayobami 27 Nov '11

27 Nov '11

I have my Freeswitch box connected to some other computer through wireless, FS automatically detects the IP address of the wireless adapter and things work fine every soft phone could register and calls were routed appropriately. But when I connected the Freeswitch box to a switch with the other computers many issues came up. At first freeswitch does not bind to the ethernet adapter, it indicates that its binding to the loopback IP address 127.0.0.1, so I tried restarting several times, no luck, so I decided to do some tweaking in Vars.xml I changed this <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> to <X-PRE-PROCESS cmd="set" data="domain=10.0.0.10"/> and in the internal.xml in sip_profile folder I changed these two settings from default values to so after all these changes, FS now binds to the ethernet adapter. But the problem now is no softphone on other computers could register again, they are all giving error 408 which means they could not register, I have googled and looked for solutions, but couldnot find any -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Soft-phones-could-not-registe… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 0

Sofia late-negotiation on re-INVITE (codec-modification)
by Phil Quesinberry 27 Nov '11

27 Nov '11

I think I might be experiencing a similar problem here... A friend of mine dialed into one of my conference rooms with only the GSM codec enabled on his mobile SIP client (Linphone). Despite the fact that I offered GSM as one of the codecs, FS only appeared to offer G711u and the call failed. When he set G711u as an option on his end, he was able to connect. Late negotiation is enabled. Some relevant settings: internal.xml: <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="inbound-codec-negotiation" value="greedy"/> vars.xml: <X-PRE-PROCESS cmd="set" data="inbound_codec_prefs=PCMU@20i,PCMU,PCMA,GSM,ILBC"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU@20i,PCMU,PCMA,GSM,ILBC"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@20i,PCMU,PCMA,GSM,ILBC"/> SIP transaction info follows: ------------------------------------------------------------------------ recv 765 bytes from udp/[68.47.xx.xx]:5060 at 06:59:23.381436: ------------------------------------------------------------------------ INVITE sip:3030@myswitch.org:5080 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.15:5060;rport;branch=z9hG4bK1252688933 From: "resist0r" <sip:resist0r@68.47.xx.xx>;tag=539248952 To: <sip:3030@myswitch.org:5080> Call-ID: 1476867094 CSeq: 20 INVITE Contact: <sip:resist0r@68.47.xx.xx> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.4.3 (eXosip2/3.1.0) Subject: Phone call Expires: 120 Content-Length: 234 v=0 o=resist0r 1856 1856 IN IP4 68.47.xx.xx s=Talk c=IN IP4 68.47.xx.xx t=0 0 m=audio 9003 RTP/AVP 0 0 3 101 a=rtpmap:0 GSM/22050 a=rtpmap:0 GSM/11025 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 ------------------------------------------------------------------------ send 354 bytes to udp/[68.47.xx.xx]:5060 at 06:59:23.381660: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.1.15:5060;rport=5060;branch=z9hG4bK1252688933;received=68.47.xx.xx From: "resist0r" <sip:resist0r@68.47.xx.xx>;tag=539248952 To: <sip:3030@myswitch.org:5080> Call-ID: 1476867094 CSeq: 20 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f2cf68b 2011-11-20 18-40-41 -0500 Content-Length: 0 ------------------------------------------------------------------------ 2011-11-25 01:59:23.365421 [NOTICE] switch_channel.c:920 New Channel sofia/external/resist0r(a)68.47.xx.xx [b9d2afd7-e0ba-4168-acee-29ccd2dc4791] 2011-11-25 01:59:23.365421 [DEBUG] sofia.c:5356 Channel sofia/external/resist0r(a)68.47.xx.xx entering state [received][100] 2011-11-25 01:59:23.365421 [DEBUG] sofia.c:5367 Remote SDP: v=0 o=resist0r 1856 1856 IN IP4 68.47.xx.xx s=Talk c=IN IP4 68.47.xx.xx t=0 0 m=audio 9003 RTP/AVP 0 0 3 101 a=rtpmap:0 GSM/11025 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[PCMU:0:8000:20:64000] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[PCMU:0:8000:20:64000] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[PCMA:8:8000:20:64000] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[GSM:3:8000:20:13200] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [GSM:0:11025:20:64000]/[iLBC:97:8000:30:13330] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:4755 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-11-25 01:59:23.365421 [DEBUG] sofia_glue.c:2869 Set Codec sofia/external/resist0r(a)68.47.xx.xx PCMU/8000 20 ms 160 samples 64000 bits Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

2 1

Skypeopen dont send caller audio
by JoZu 27 Nov '11

27 Nov '11

Hi, i have a little problem i have FS installed, skypeopen_oss client and skypeopen.ko oss sound driver installed. I configure in public a DID number bridge to skypopen/RR/skype22, the call going from DID to SKYPE1 and call SKYPE22, here is the LOG: http://pastebin.freeswitch.org/17877 http://pastebin.freeswitch.org/17877 All work OK, but the sound from DID caller to Skype22 dont reach, no sound, Skype22 to DID caller works ok. Questions: Skype_oss and the skypeopen.ko require a sound card to work? What is the problem? Codecs? Please helpme, thanks and sorry my english. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Skypeopen-dont-send-caller-au… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 0

No shared modules built, linking problems on NetBSD
by Jesse Peterson 27 Nov '11

27 Nov '11

Hello, I'm having some problems building FreeSWITCH head from Git on NetBSD. There are some known NetBSD issue in the source that need to be corrected for but once corrected then the compilation happens successfully. However once installed FreeSWITCH has only installed .a and .la files into /usr/local/freeswitch/mod/ instead of .so files. Thus when FreeSWITCH starts it cannot load any of the modules because there's no dynamic libraries. I don't know if it's related but when building modules and then linking modules many of them have: *** Warning: This system can not link to static lib archive libs/apr-util/libaprutil-1.la. *** I have the capability to make that library automatically link in when *** you link to this library. But I can only do this if you have a *** shared version of the library, which you do not appear to have. *** Warning: This system can not link to static lib archive /home/local-src/freeswitch/libs/apr-util/xml/expat/lib/libexpat.la. *** I have the capability to make that library automatically link in when *** you link to this library. But I can only do this if you have a *** shared version of the library, which you do not appear to have. *** Warning: Trying to link with static lib archive libs/libedit/src/.libs/libedit.a. *** I have the capability to make that library automatically link in when *** you link to this library. But I can only do this if you have a *** shared version of the library, which you do not appear to have *** because the file extensions .a of this argument makes me believe *** that it is just a static archive that I should not use here. These vary by the specific module. This system is NetBSD/i386 5.1 with LibTool 2.2.6b with autoconf 2.68 and automake 1.11.1 (all from pkgsrc). It appears there was one shared library built: ./libs/libfreeswitch.so so it seems it *can* build shared libraries, just none of the modules perhaps? Any ideas or help would be appreciated. Thanks! - Jesse

1 0

capture not working on freeswitch.org?
by David Wafula 26 Nov '11

26 Nov '11

I wanted to create an account on freeswitch.org, because i need to use pastebin. That is how to get the credentials for pastebin, right? But i am unable register there, the capture is not working. regards, -- David Wafula

4 5

Timing issues in AWS?
by Michael Picher 26 Nov '11

26 Nov '11

Hi guys, Trying to get to the bottom of some conference bridge issues I'm having with running the system in AWS. We're hearing a bunch of snap-crackle-pops in conference bridges and when I tcpdum on the server itself I see them in the RTP and see RTP timestamp problems. I've run the following: timer_test freeswitch@127.0.0.1@internal> timer_test 120 10 Avg: 120.004ms Total Time: 1200.315ms 2011-11-16 10:25:56.121131 [CONSOLE] mod_commands.c:461 Timer Test: samplecount after init: 1 freeswitch@127.0.0.1@internal> 2011-11-16 10:25:56.219163 [CONSOLE] mod_commands.c:466 Timer Test: samplecount after first step: 2 2011-11-16 10:25:56.339195 [CONSOLE] mod_commands.c:475 Timer Test: 1 sleep 120 120006 2011-11-16 10:25:56.459259 [CONSOLE] mod_commands.c:475 Timer Test: 2 sleep 120 120040 2011-11-16 10:25:56.579259 [CONSOLE] mod_commands.c:475 Timer Test: 3 sleep 120 119976 2011-11-16 10:25:56.699291 [CONSOLE] mod_commands.c:475 Timer Test: 4 sleep 120 120007 2011-11-16 10:25:56.819318 [CONSOLE] mod_commands.c:475 Timer Test: 5 sleep 120 120001 2011-11-16 10:25:56.939375 [CONSOLE] mod_commands.c:475 Timer Test: 6 sleep 120 120031 2011-11-16 10:25:57.059397 [CONSOLE] mod_commands.c:475 Timer Test: 7 sleep 120 119996 2011-11-16 10:25:57.179422 [CONSOLE] mod_commands.c:475 Timer Test: 8 sleep 120 119994 2011-11-16 10:25:57.299461 [CONSOLE] mod_commands.c:475 Timer Test: 9 sleep 120 120005 2011-11-16 10:25:57.419478 [CONSOLE] mod_commands.c:475 Timer Test: 10 sleep 120 119991 test_time freeswitch@127.0.0.1@internal> time_test 600 10 test 1 sleep 600 1592 test 2 sleep 600 986 test 3 sleep 600 1018 test 4 sleep 600 980 test 5 sleep 600 1005 test 6 sleep 600 1000 test 7 sleep 600 972 test 8 sleep 600 990 test 9 sleep 600 1006 test 10 sleep 600 994 avg 1054 For kernel: [root@openuc bin]# uname -r 2.6.21.7-2.fc8xen CONFIG_HZ: [root@openuc bin]# grep CONFIG_HZ /boot/config-* /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ_100=y /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_250 is not set /boot/config-2.6.16.33-xenU-x86_64:# CONFIG_HZ_1000 is not set /boot/config-2.6.16.33-xenU-x86_64:CONFIG_HZ=100 /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_100 is not set /boot/config-2.6.18-164.15.1.el5.centos.plus:# CONFIG_HZ_250 is not set /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ_1000=y /boot/config-2.6.18-164.15.1.el5.centos.plus:CONFIG_HZ=1000 /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_100 is not set /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_250 is not set /boot/config-2.6.21-2952.fc8xen:# CONFIG_HZ_300 is not set /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ_1000=y /boot/config-2.6.21-2952.fc8xen:CONFIG_HZ=1000 are the xenU kernel settings screwing me here? Thanks, Mike -- There are 10 kinds of people in this world, those who understand binary and those who don't. mpicher(a)gmail.com blog: http://www.sipxecs.info call: sip:mpicher@sipxecs.info

8 15