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Freeswitch with Sipgate
by realdoe 19 May '11

19 May '11

Hi all, I try to get Freeswitch running with Sipgate as a Gateway but when I enter "sofia status" I get the following error: 2011-05-19 15:24:06.897915 [ERR] sofia_reg.c:1611 sipconnect.sipgate.de Registration Failed with status Forbidden (client configuration error) [403]. failure #4 I created a file called sipgate.xml under /opt/freeswitch/conf/sip_profiles/external/ and filled it with the following: include gateway name="sipgate.de" param name="proxy" value="sipconnect.live.sipgate.de"/ param name="from-domain" value="sipgate.de"/ param name="username" value="MY_NAME"/ param name="password" value="MY_PASSWORD"/ /gateway /include (Removed the brackets...) Does anyone know what I might be missing or what went wrong? Thanks for your time... Greetings Realdoe -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-with-Sipgate-tp638… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Spoofed CID?
by mazilo 19 May '11

19 May '11

Today, after my phone rang with a CID of (443)366-9222, I took a look at /var/log/freeswitch/freeswitch.log file on my Seagate DockStar (excerpted below). What caught my attention is the last line which shows two numbers, namely 4433669222 and 4016488312. The question I have is if 4433669222 is a spoofed number? 2011-05-18 12:54:03.258849 [NOTICE] switch_channel.c:812 New Channel sofia/external/4433669222(a)64.154. 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4744 Channel sofia/external/4433669222(a)64.154.41.150 enteri 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4755 Remote SDP: v=0^M o=root 241964608 241964608 IN IP4 64.154.41.150^M s=Asterisk PBX 1.6.2.13^M c=IN IP4 64.154.41.150^M t=0 0^M m=audio 10730 RTP/AVP 0 8 3 18 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:3 GSM/8000^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=ptime:20^M 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722: 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU: 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:2760 Set Codec sofia/external/4433669222(a)64.154.41.150 2011-05-18 12:54:03.258849 [DEBUG] sofia_glue.c:4751 Set 2833 dtmf send/recv payload to 101 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4922 (sofia/external/4433669222(a)64.154.41.150) State Change 2011-05-18 12:54:03.258849 [DEBUG] sofia.c:4922 (sofia/external/4433669222(a)64.154.41.150) State Change 2011-05-18 12:54:03.258849 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/4433669222@64 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:320 (sofia/external/4433669222(a)64.154.4 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:356 (sofia/external/4433669222(a)64.154.4 2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:84 sofia/external/4433669222(a)64.154.41.150 SOFIA INIT 2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:124 (sofia/external/4433669222(a)64.154.41.150) State Cha 2011-05-18 12:54:03.265631 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/4433669222@64 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:356 (sofia/external/4433669222(a)64.154.4 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:320 (sofia/external/4433669222(a)64.154.4 2011-05-18 12:54:03.265631 [DEBUG] switch_channel.c:1668 (sofia/external/4433669222(a)64.154.41.150) Cal 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:359 (sofia/external/4433669222(a)64.154.4 2011-05-18 12:54:03.265631 [DEBUG] mod_sofia.c:147 sofia/external/4433669222(a)64.154.41.150 SOFIA ROUTI 2011-05-18 12:54:03.265631 [DEBUG] switch_core_state_machine.c:77 sofia/external/4433669222(a)64.154.41. 2011-05-18 12:54:03.265631 [INFO] mod_dialplan_xml.c:331 Processing 4433669222 <4433669222>->401648831 ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Spoofed-CID-tp6380373p6380373… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] G729 on windows???
by mazilo 19 May '11

19 May '11

Mogsy.uk wrote: > Is there a G729 codec available yet for windows? I am just wondering some free windows softphones have a support for G729 CoDec. Can anyone make use this CoDec on Windows FS? ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/G729-on-windows-tp5213277p638… Sent from the freeswitch-users mailing list archive at Nabble.com.

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T.38 via UPDATE request
by Peter P GMX 19 May '11

19 May '11

Hello, we have the following problem. 2 Fax machines are communicating via Freeswitch. One is externally attached via a Telco who is able to handle T.38. The other one is attached locally. When 2 Fax machines start syncing each other, the Telco sends a SIP UPDATE message with T.38 SDP, as it detects fax during the fax negociations. Freeswitch answers with an SIP OK message back to the telco, and I can see the T.38 SDP on the debug console of freeswitch. Then nothing happens any more until one of fax machines detects timeout. We have set proxy-media to true. However is was done during call setup when both machines communicated with G711 SDP. The UPDATE message was commited by FS to the telco, but was not sent to the other fax, so I think in this case Freeswitch is supposed to transcode between T.38 and G711 which it cannot do, as we know. How can I overcome this scenario? Is this a defect, should freeswitch send the UPDATE message to the other fax? Or is there a workaround? Best regards Peter Here's the UPDATE message: UPDATE sip:mod_sofia@82.115.xx.xxx:5080 SIP/2.0. Call-ID: 5283fe4e-334f-122d-d1b9-001517956764. Contact: <sip:212.91.xxx.xxx:5060>. Content-Type: application/sdp. CSeq: 16340063 UPDATE. From: <sip:06912345678@sip.telco.de>;tag=00-08135-017e041a-4d21f6037. Max-Forwards: 31. Route: <sip:82.115.96.165;lr;ftag=pH663F4S02erm>. To: "030987654321" <sip:030987654321@82.115.xx.xxx>;tag=pH663F4S02erm. User-Agent: Cirpack/v4.41e (gw_sip). Via: SIP/2.0/UDP 212.91.xxx.xxx:5060;branch=z9hG4bK-178C-1923B31. Content-Length: 300. . v=0. o=cp10 125551618103 125551618105 IN IP4 212.91.xxx.xx. s=SIP Call. c=IN IP4 212.91.xxx.xx. t=0 0. m=image 6860 udptl t38. a=sendrecv. a=T38FaxVersion:0. a=T38MaxBitRate:9600. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:1000. a=T38FaxMaxDatagram:200. a=T38FaxUdpEC:t38UDPRedundancy.

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Something as powerful as voip.ms?
by Sidharth Kshatriya 19 May '11

19 May '11

Ok guys, a few days ago I had asked people for a good round voip provider and people pitched in with a lot of names. Among all of them, I checked out voip.ms and I am totally impressed by the hundreds of options in their login interface. Thats what really set it apart from others -- for me. Many other providers seemed to have similar pricing but I haven't found any website that gives that kind of login interface options + information transparency that voip.ms does. I really want to sign up for them. However there is a catch. They don't accept Paypal / Credit cards from India!! This is unusual and strange as I've never had any problems before. The only other route to pay voip.ms is western union in which I need to pay in $100 increments or Bank transfers in $500 increments. Needless to say, I'd like to avoid all that. I am considering abandoning voip.ms but just can't seem to find any other provider which are "equivalent" in terms of flexibility and power. Any help? Suggestions? I'm looking for inbound DID services only (to run an IVR) * One US number non-toll free 5+ channels * One US toll free number * (Maybe) UK DID / UK toll free DID Thanks, Sidharth -- Sidharth Kshatriya www.sidk.info

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Re: [Freeswitch-users] vm_list sends Notify to phone
by Bill Dunn - VCI Internet Services 18 May '11

18 May '11

I found an answer to my problem at http://www.voip-info.org./wiki/view/Polycom+SoundPoint+IP+MWI+audio Bill Dunn ----- Original Message ----- From: Bill Dunn - VCI Internet Services To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, May 18, 2011 9:17 AM Subject: vm_list sends Notify to phone I am using the API. When I issue a "vm_list phone#@domain" command Freeswitch sends a Notify packet to the phone. My Polycom phones give me a audible alert each time they receive one of these. Is there a way to stop Freeswitch from sending a Notify packet to the phone each time a "vm_list" command is sent to the API? Bill Dunn

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ZRTP of freeswitch.
by Jakson Kalsson 18 May '11

18 May '11

Does freeswitch support ZRTP? Thansk -- havesoftware, Inc. http://www.havesoftware.com Jakson Kalsson Senior Programmer jakkalsoon(a)havesoftware.com

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Voicemail macro error in log
by Joseph L. Casale 18 May '11

18 May '11

While looking at something, I noticed a [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_ack]: 'saved' did not match any patterns pass by but this match exists in the voicemail_ack macro. I looked in freeswitch.xml.fsxml and it is being loaded. What could cause this? Thanks, jlc

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FreeSWITCH Conference Call Today
by Michael Collins 18 May '11

18 May '11

Hello all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_05_18 It's a bit light but there are a few things to talk about. Please bring your questions for discussion. Thanks, MC

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FreeSwitch Compile Error
by vivid 18 May '11

18 May '11

Can anyone tell me how to solve this problem? ./configure make /////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// Compiling src/switch_apr.c ... Compiling src/switch_buffer.c ... Compiling src/switch_caller.c ... Compiling src/switch_channel.c ... Compiling src/switch_console.c ... Compiling src/switch_mprintf.c ... Compiling src/switch_core_media_bug.c ... Compiling src/switch_core_timer.c ... Compiling src/switch_core_asr.c ... Compiling src/switch_core_event_hook.c ... Compiling src/switch_core_speech.c ... Compiling src/switch_core_memory.c ... Compiling src/switch_core_codec.c ... Compiling src/switch_core_file.c ... Compiling src/switch_core_hash.c ... Compiling src/switch_core_sqldb.c ... cc1: warnings being treated as errors src/switch_core_sqldb.c: In function '_switch_cache_db_get_db_handle': src/switch_core_sqldb.c:314:47: error: comparison between 'switch_odbc_status_t' and 'enum <anonymous>' src/switch_core_sqldb.c: In function 'switch_cache_db_execute_sql_real': src/switch_core_sqldb.c:389:90: error: comparison between 'switch_status_t' and 'enum <anonymous>' make[1]: *** [libfreeswitch_la-switch_core_sqldb.lo] Error 1 make: *** [all] Error 2

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