August 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

make && make install error
by Ikenna Onyeagba 20 Aug '12

20 Aug '12

i get this error when i run the make && make install error. Making all in doc make[3]: Entering directory `/home/free/usr/src/freeswitch-1.0.6/libs/libedit/doc' if test "mdoc" = "mdoc"; then\ cp ./editline.3.roff editline.3;\ else\ gawk -f ./mdoc2man.awk ./editline.3.roff > editline.3 || rm -f editline.3;\ fi; if test "mdoc" = "mdoc"; then\ cp ./editrc.5.roff editrc.5;\ else\ gawk -f ./mdoc2man.awk ./editrc.5.roff > editrc.5 || rm -f editrc.5;\ fi; make[3]: Leaving directory `/home/free/usr/src/freeswitch-1.0.6/libs/libedit/doc' make[3]: Entering directory `/home/free/usr/src/freeswitch-1.0.6/libs/libedit' make[3]: Leaving directory `/home/free/usr/src/freeswitch-1.0.6/libs/libedit' make[2]: Leaving directory `/home/free/usr/src/freeswitch-1.0.6/libs/libedit' make[1]: Leaving directory `/home/free/usr/src/freeswitch-1.0.6/libs/libedit' gcc -E /home/free/usr/src/freeswitch-1.0.6/src/include/switch_cpp.h -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE$x$=x -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD$x$=x 2>/dev/null | grep -v "^#" > src/include/switch_swigable_cpp.h make: *** [src/include/switch_swigable_cpp.h] Error 1 [root@Ikenna-TOSH freeswitch-1.0.6]# pls i need help on how to resolve this problem urgently. thanks in advance Ikenna

4 3

CIDlookup
by Brett Wilson 20 Aug '12

20 Aug '12

I am trying to get CID lookup to display on the phone's display. I am using this code in the beginning of the public.xml dialplan: <extension name="cid_number_cleanup" continue="true"> <condition field="caller_id_number" expression="^(?:\+)(\d+)$"> <action application="set" data="effective_caller_id_number=$1" inline="true"/> </condition> </extension> <extension name="cid_name_cleanup" continue="true"> <condition field="caller_id_name" expression="^(?:\+)(\d+)$"> <action application="set" data="effective_caller_id_name=$1" inline="true"/> </condition> </extension> <extension name="cid_lookup-country_code_1" continue="true"> <condition field="${module_exists(mod_cidlookup)}" expression="true"/> <condition field="caller_id_name" expression="^(?:\+)(\d+)$|^$"/> <condition field="caller_id_number" expression="^(?:\+1|1)?([2-9]\d\d[2-9]\d{6})$"> <action application="cidlookup" data="$1"/> </condition> </extension> The lookup works - I can see the result in the cdr records. But it does not display on the phone's display. When we get a call in with actual cnam info, that does display on the phone. The difference I think is when we get a cnam call, sip_from_display is filled with the name. When we do the cid lookup, it is not filled. I tried to set sip_from_display myself through channel variables in the bridge string as well as export but it is not working. Any ideas on how I can do this? Being able to control what is displayed on the screen would be ideal so I can let users know when they are getting a call direct to their extension or an ACD call. TIA. ******************************************* Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: <mailto:Brett.Wilson@launch3.net> Brett.Wilson(a)launch3.net AOL IM: <mailto:Brett.Wilson@launch3.net> Brett.Wilson(a)launch3.net <http://www.launch3.net/> www.Launch3.net <http://www.launch3telecom.com/> www.Launch3telecom.com ******************************************* <http://launch3telecom.blogspot.com/> Description: Description: Description: Blogger-logo <http://www.facebook.com/pages/Launch-3-Telecom/165610820178720> Description: Description: Description: FaceBook-Logo <https://twitter.com/launch3telecom> Description: Description: Description: Twitter-Logo <https://plus.google.com/102713212642017290959> Description: Description: Description: GPlus-Logo

1 0

Mod_callcenter - cannot blind transfer 1 legged calls
by Brett Wilson 20 Aug '12

20 Aug '12

Hey guys, I'm trying to implement mod callcenter, I've got everything setup (from within fusionpbx) and I'm getting 'cannot blind transfer 1 legged calls' when trying to send a call to the acd. Log: http://pastebin.com/y2U1g2f2 Also, an unrelated question - what is the normal way to set up a fs box behind a firewall and on a local lan with regards to hostname and such? I have the box setup on the dhcp server as always getting assigned ip 192.168.2.7 via mac address. I prefer this to setting static ip in the host os. The box is sip registering to my voice provider. On the dhcp server, I am also running a dns server that resolves some local lan hostnames and forwards anything else to google or opendns. Also computer hostnames are automatically added to the dns database, so if the hostname for the fs box is 'hades', 'hades.l3office.net' will resolve to the fs box, but only from within the lan. L3office.net does not actually exist and is not routable from the wan. Where I am unclear is what to set the domain to in the fs config, hades.l3office.net or 192.168.2.7? And in the phone's sip server? TIA for the assistance. ******************************************* Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: <mailto:Brett.Wilson@launch3.net> Brett.Wilson(a)launch3.net AOL IM: <mailto:Brett.Wilson@launch3.net> Brett.Wilson(a)launch3.net <http://www.launch3.net/> www.Launch3.net <http://www.launch3telecom.com/> www.Launch3telecom.com ******************************************* <http://launch3telecom.blogspot.com/> Description: Description: Description: Blogger-logo <http://www.facebook.com/pages/Launch-3-Telecom/165610820178720> Description: Description: Description: FaceBook-Logo <https://twitter.com/launch3telecom> Description: Description: Description: Twitter-Logo <https://plus.google.com/102713212642017290959> Description: Description: Description: GPlus-Logo

1 0

Phantom tone with session:read()
by Neil Patel 20 Aug '12

20 Aug '12

I'm playing a wav file via lua app and there is a phantom tone that plays at a certain part of the file. It's not there when I play the file on my PC natively (Audacity, Quicktime, etc.), but it's there when I play it with session:read(). Any suggestions on how to debug this? Thanks, Neil

1 0

How to config FS to listen to early media at 180?
by Pip Live 20 Aug '12

20 Aug '12

Hi, my FS for outbound PSTN gateway is configed as: (in sofia.conf xml) <param name="inbound-bypass-media" value="false"/> <param name="inbound-late-negotiation" value="true"/> ..... (in particular gateway.conf xml) <action application="set" data="ignore_early_media=false"/> <action application="set" data="bridge_early_media=true"/> <action application="bridge" data="sofia/gateway/PSTN1/$1"/> But outbound call can not hear the early media played by destination number. I checked Wiki which says FS will connect media when destination send 183, and 180 means end-points should generate the faked ring tone; but in part of Europ (Germany) & Asia countries, PSTN operators send early media with 180. Checked logger, it did showed a response of 180. So is there anyway to config FS to connect media at 180 response from destination? Rgds, Pip

4 3

mobile diallers
by clive engelberg 20 Aug '12

20 Aug '12

Hi Someone recommended http://www.acrobits.cz Worth a try. regards Clive On Fri, 17 Aug 2012 11:46:18 +0400 freeswitch-users-request(a)lists.freeswitch.org wrote IMSDroid is based on Doubango developed by Doubango Telecom, which also has support for iPhone, Windows and Mac. You should check their website at http://www.doubango.org [1]. I think they may provide you commercial support for their productions as well, just talk to them. Thank you. On Fri, Aug 17, 2012 at 9:05 AM, vaad.fabi(a)gmail.com [2] wrote: IMSDROID - only Android platform, no centralizied dev teamvendor with support and SLA SIPDROID - only Android platform, no centralizied dev teamvendor with support and SLA Counterpath - as i know no mobile platforms Searching... On 08/16/2012 08:58 PM, freeswitch wrote: IMSDROID will be good! At 2012-08-16 20:44:54,BookBag [4] wrote: I'm surprised no one said Bria by counterpath. Which allows for xmpp and presence. The softphone allows you to send sms also if you buy the full version. Another one that I tested is Csipsimple, this one allows sms for free. But my absolute favorite is Zoiper, unfortunately no sms capability but wasnt buggy except for one bug I encountered with sending the correct dtmf style. Thank god their new update took care of that. Phone now seems very solid and now includes a QR reader. On Thu, Aug 16, 2012 at 6:16 AM, vaad.fabi(a)gmail.com [5] wrote: Does they provide these software development, customisation and support ? i think no. I need vendordevelopment team who can provide developmentcustomisationsupport. Anyway thx. On 08/16/2012 12:53 PM, Mohammad Emran wrote: > Then try nimbuzz or fring.... If it works for u. > > Sent from my iPhone > > On Aug 16, 2012, at 3:42 PM, "vaad.fabi(a)gmail.com [7]" wrote: > >> Seems to be ok for a simple dialer, but i forgot to mention - i need >> advanced options for dialers, such as: IM (xmppfacebooktwitter >> integration), video calls, balance state, check direction rate, >> callback, sms, top up and etc. It's smth like UC application with >> calling support as one of the many options. >> >> >> On 08/16/2012 12:22 PM, Mohammad Emran wrote: >>> Try www.sipmobiledialer.com [9] >>> >>> >>> Sent from my iPad >>> >>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi(a)gmail.com [10]" wrote: >>> >>>> Hi all, >>>> >>>> Please suggest mobile voip dialer vendor for >>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms. >>>> don't suggest google plz >>>> >>>> -- >>>> Best Regards, >>>> Vadim F. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org [12] >>>> http://www.freeswitchsolutions.com [13] >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com [14] >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org [15] >>>> http://wiki.freeswitch.org [16] >>>> http://www.cluecon.com [17] >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users(a)lists.freeswitch.org [18] >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [19] >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [20] >>>> http://www.freeswitch.org [21] >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org [22] >>> http://www.freeswitchsolutions.com [23] >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com [24] >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org [25] >>> http://wiki.freeswitch.org [26] >>> http://www.cluecon.com [27] >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org [28] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [29] >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [30] >>> http://www.freeswitch.org [31] >> >> -- >> Best Regards, >> Vadim F. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org [32] >> http://www.freeswitchsolutions.com [33] >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com [34] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [35] >> http://wiki.freeswitch.org [36] >> http://www.cluecon.com [37] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org [38] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [39] >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [40] >> http://www.freeswitch.org [41] > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org [42] > http://www.freeswitchsolutions.com [43] > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com [44] > > Official FreeSWITCH Sites > http://www.freeswitch.org [45] > http://wiki.freeswitch.org [46] > http://www.cluecon.com [47] > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org [48] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [49] > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [50] > http://www.freeswitch.org [51] -- Best Regards, Vadim F. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org [52] http://www.freeswitchsolutions.com [53] FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com [54] Official FreeSWITCH Sites http://www.freeswitch.org [55] http://wiki.freeswitch.org [56] http://www.cluecon.com [57] FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org [58] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [59] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [60] http://www.freeswitch.org [61] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org [62] http://www.freeswitchsolutions.com [63] FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com [64] Official FreeSWITCH Sites http://www.freeswitch.org [65] http://wiki.freeswitch.org [66] http://www.cluecon.com [67] FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org [68] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [69] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [70] http://www.freeswitch.org [71] -- Best Regards, Vadim F. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org [72] http://www.freeswitchsolutions.com [73] FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com [74] Official FreeSWITCH Sites http://www.freeswitch.org [75] http://wiki.freeswitch.org [76] http://www.cluecon.com [77] FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org [78] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [79] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [80] http://www.freeswitch.org [81] -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com [82] Email: shaheryarkh(a)googlemail.com [83] Links: ------ [1] http://www.doubango.org [2] mailto:vaad.fabi@gmail.com [3] mailto:vaad.fabi@gmail.com [4] mailto:asaad2@gmail.com [5] mailto:vaad.fabi@gmail.com [6] mailto:vaad.fabi@gmail.com [7] mailto:vaad.fabi@gmail.com [8] mailto:vaad.fabi@gmail.com [9] http://www.sipmobiledialer.com [10] mailto:vaad.fabi@gmail.com [11] mailto:vaad.fabi@gmail.com [12] mailto:consulting@freeswitch.org [13] http://www.freeswitchsolutions.com [14] http://www.cudatel.com [15] http://www.freeswitch.org [16] http://wiki.freeswitch.org [17] http://www.cluecon.com [18] mailto:FreeSWITCH-users@lists.freeswitch.org [19] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [20] http://lists.freeswitch.org/mailman/options/freeswitch-users [21] http://www.freeswitch.org [22] mailto:consulting@freeswitch.org [23] http://www.freeswitchsolutions.com [24] http://www.cudatel.com [25] 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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [50] http://lists.freeswitch.org/mailman/options/freeswitch-users [51] http://www.freeswitch.org [52] mailto:consulting@freeswitch.org [53] http://www.freeswitchsolutions.com [54] http://www.cudatel.com [55] http://www.freeswitch.org [56] http://wiki.freeswitch.org [57] http://www.cluecon.com [58] mailto:FreeSWITCH-users@lists.freeswitch.org [59] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [60] http://lists.freeswitch.org/mailman/options/freeswitch-users [61] http://www.freeswitch.org [62] mailto:consulting@freeswitch.org [63] http://www.freeswitchsolutions.com [64] http://www.cudatel.com [65] http://www.freeswitch.org [66] http://wiki.freeswitch.org [67] http://www.cluecon.com [68] mailto:FreeSWITCH-users@lists.freeswitch.org [69] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [70] http://lists.freeswitch.org/mailman/options/freeswitch-users [71] http://www.freeswitch.org [72] mailto:consulting@freeswitch.org [73] http://www.freeswitchsolutions.com [74] http://www.cudatel.com [75] http://www.freeswitch.org [76] http://wiki.freeswitch.org [77] http://www.cluecon.com [78] mailto:FreeSWITCH-users@lists.freeswitch.org [79] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [80] http://lists.freeswitch.org/mailman/options/freeswitch-users [81] http://www.freeswitch.org [82] mailto:shari_786pk@hotmail.com [83] mailto:shaheryarkh@googlemail.com ____________________________________________________________ South Africas premier free email service - www.webmail.co.za For super low premiums, click here. http://www.dialdirect.co.za/campaign/?vdn=15828

4 5

mod_callcenter record template
by Vbvbrj 20 Aug '12

20 Aug '12

In mod callcenter.conf.xml there is: <param name="record-template" value="$${base_dir}/recordings/sales/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/> This names the file with callers number and callcenter's assigned number. Bit how to include in the file name the agent's number, which responded? Which variable to use?

1 0

Recording problem
by Vbvbrj 20 Aug '12

20 Aug '12

Hello. I have a problem with recorded files (wav, mp3, gsm) there are three agents in callcenter which receives calls from same external provider. The problem is that recorded sound from one agent is good quality, while other two recorded sounds have dropped samples and its like the sound is at 2x speed. In vars.xml I have: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@32000h@20i,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> I am using mod_callcenter. There are 3 agents: user 892 at D-Link DPH-150S user 743 at D-Link DPH-150S user 897 at Linksys PAP2T There is incoming calls from same external number (multiple channels), the log is: ..... a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.658288 [DEBUG] sofia.c:6060 Remote SDP: a5079181-e072-4eee-9291-95a4ab40eabd v=0 a5079181-e072-4eee-9291-95a4ab40eabd o=- 1345205895 1345205895 IN IP4 192.168.0.6 a5079181-e072-4eee-9291-95a4ab40eabd s=- a5079181-e072-4eee-9291-95a4ab40eabd c=IN IP4 192.168.0.6 a5079181-e072-4eee-9291-95a4ab40eabd t=0 0 a5079181-e072-4eee-9291-95a4ab40eabd m=audio 13630 RTP/AVP 8 0 a5079181-e072-4eee-9291-95a4ab40eabd a=rtpmap:8 PCMA/8000 a5079181-e072-4eee-9291-95a4ab40eabd a=rtpmap:0 PCMU/8000 a5079181-e072-4eee-9291-95a4ab40eabd a=silenceSupp:off - - - - a5079181-e072-4eee-9291-95a4ab40eabd ..... a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMA:8:8000:20:64000]/[SPEEX:99:32000:20:44000] a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] sofia_glue.c:3027 Set Codec sofia/orange/extern_number@192.168.0.5:5061 PCMA/8000 20 ms 160 samples 64000 bits a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] switch_core_codec.c:111 sofia/orange/extern_number@192.168.0.5:5061 Original read codec set to PCMA:8 After playing greeting, caller is routed to callcenter. When an agent answers the log has: For user 892: 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia.c:6060 Remote SDP: 55969678-ab9f-4007-8023-304f4b602c97 v=0 55969678-ab9f-4007-8023-304f4b602c97 o=710 38421252 24417309 IN IP4 132.101.208.92 55969678-ab9f-4007-8023-304f4b602c97 s=A conversation 55969678-ab9f-4007-8023-304f4b602c97 c=IN IP4 132.101.208.92 55969678-ab9f-4007-8023-304f4b602c97 t=0 0 55969678-ab9f-4007-8023-304f4b602c97 m=audio 10154 RTP/AVP 9 0 8 101 55969678-ab9f-4007-8023-304f4b602c97 a=rtpmap:9 G722/16000 55969678-ab9f-4007-8023-304f4b602c97 a=rtpmap:0 PCMU/8000 55969678-ab9f-4007-8023-304f4b602c97 a=rtpmap:8 PCMA/8000 55969678-ab9f-4007-8023-304f4b602c97 a=rtpmap:101 telephone-event/8000 55969678-ab9f-4007-8023-304f4b602c97 a=fmtp:101 0-15 ...... 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [G722:9:16000:20:64000]/[SPEEX:99:32000:20:44000] 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [G722:9:16000:20:64000]/[G722:9:8000:20:64000] 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [G722:9:16000:20:64000]/[PCMU:0:8000:20:64000] 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [G722:9:16000:20:64000]/[PCMA:8:8000:20:64000] 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [G722:9:16000:20:64000]/[GSM:3:8000:20:13200] 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:5097 Substituting codec G722@20i@8000h 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/sip:892@132.101.208.92:5060 G722/8000 20 ms 160 samples 64000 bits 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:892@132.101.208.92:5060 Original read codec set to G722:9 For user 743: acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] sofia.c:6060 Remote SDP: acb5bf6a-4e79-46a4-a32b-420b5301d6ec v=0 acb5bf6a-4e79-46a4-a32b-420b5301d6ec o=1743 26961311 21406313 IN IP4 132.101.207.43 acb5bf6a-4e79-46a4-a32b-420b5301d6ec s=A conversation acb5bf6a-4e79-46a4-a32b-420b5301d6ec c=IN IP4 132.101.207.43 acb5bf6a-4e79-46a4-a32b-420b5301d6ec t=0 0 acb5bf6a-4e79-46a4-a32b-420b5301d6ec m=audio 10092 RTP/AVP 0 8 101 acb5bf6a-4e79-46a4-a32b-420b5301d6ec a=rtpmap:0 PCMU/8000 acb5bf6a-4e79-46a4-a32b-420b5301d6ec a=rtpmap:8 PCMA/8000 acb5bf6a-4e79-46a4-a32b-420b5301d6ec a=rtpmap:101 telephone-event/8000 acb5bf6a-4e79-46a4-a32b-420b5301d6ec a=fmtp:101 0-15 ...... acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:20:64000]/[SPEEX:99:32000:20:44000] acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/sip:743@132.101.207.43:5060 PCMU/8000 20 ms 160 samples 64000 bits acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:743@132.101.207.43:5060 Original read codec set to PCMU:0 For user 897: 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia.c:6060 Remote SDP: 35c3b541-c78e-4d4d-96fe-f15509dc7f91 v=0 35c3b541-c78e-4d4d-96fe-f15509dc7f91 o=- 1715257 1715257 IN IP4 132.101.202.2 35c3b541-c78e-4d4d-96fe-f15509dc7f91 s=- 35c3b541-c78e-4d4d-96fe-f15509dc7f91 c=IN IP4 132.101.202.2 35c3b541-c78e-4d4d-96fe-f15509dc7f91 t=0 0 35c3b541-c78e-4d4d-96fe-f15509dc7f91 m=audio 16444 RTP/AVP 0 100 101 35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=rtpmap:0 PCMU/8000 35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=rtpmap:100 NSE/8000 35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=fmtp:100 192-193 35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=rtpmap:101 telephone-event/8000 35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=fmtp:101 0-15 35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=ptime:30 ...... 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:30:64000]/[SPEEX:99:32000:20:44000] 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:30:64000]/[G722:9:8000:20:64000] 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000] 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMA:8:8000:20:64000] 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:30:64000]/[GSM:3:8000:20:13200] 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:5097 Substituting codec PCMU@30i@8000h 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/sip:897@132.101.202.2:5060 PCMU/8000 30 ms 240 samples 64000 bits 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:897@132.101.202.2:5060 Original read codec set to PCMU:0 See the difference for users 892 and 743. For user 892 a substitution for codecs is done, while for user 743 is not. Both users are registered from the same model of IP phone with same configs except sip configuration for login. What can be wrong? Thank you.

3 7

CALL TIMEOUT and interprise bridge
by Sean Devoy 19 Aug '12

19 Aug '12

Hi, I have noticed that "SET [call_timeout]=[25]" seems to be ignored when using an Enterprise bridge (:_:) to two extensions. The timeout is always 60 seconds. Do I just need to set the timeout in {} on each leg? Sean

3 2

mod_spy codec issue
by Willoughby, Ian 19 Aug '12

19 Aug '12

I'm on the latest build and have a problem with garbled audio while executing userspy. When I do a show channels, I see that my read_codec is set to L16 although I don't have that set if the global coded prefs. Is there a way to force or deny codecs for userspy? I have tried the following to no avail: <action application="export" data="nolocal:absolute_codec_string=PCMU,PCMA"/> <action application="export" data="absolute_codec_string=PCMU,PCMA"/> <action application="answer"/> <action application="set" data="domain_name=blah.com"/> <action application="userspy" data="{absolute_codec_string=PCMU}$1@${domain_name} ${hash(select/${domain_name}/$1)}"/> Ian

1 0