April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] PRI channels suspended
by Ashish gautam 16 Apr '13

16 Apr '13

2013-04-16 14:41:04.214291 [ERR] ftmod_libpri.c:1950 [s1c30][1:30] -- T316 timed out, channel reached restart attempt limit '3' and is suspended This is what happening on restarting/starting the FS for all the channels(this is only for channel number 30). How to resolve it? Please help. -- Regards. On Tue, Apr 16, 2013 at 2:46 PM, Ashish gautam <ashish(a)nms.co.in> wrote: > Sorry Michael mistakenly sent to you personally again. Sending to the list > once more. > > > On Tue, Apr 16, 2013 at 2:45 PM, Ashish gautam <ashish(a)nms.co.in> wrote: > >> 2013-04-16 14:41:04.214291 [ERR] ftmod_libpri.c:1950 [s1c30][1:30] -- >> T316 timed out, channel reached restart attempt limit '3' and is suspended >> >> This is what happening on restarting/starting the FS for all the >> channels(this is only for channel number 30). >> >> How to resolve it? Please help. >> >> -- >> Regards. >> >> On Mon, Apr 15, 2013 at 9:51 PM, Michael Collins <msc(a)freeswitch.org>wrote: >> >>> Please only send this to the list, otherwise it is nearly impossible for >>> me to reply in such a way that everyone on the list can join the >>> conversation. >>> -MC >>> >>> >>> On Sun, Apr 14, 2013 at 11:29 PM, Ashish gautam <ashish(a)nms.co.in>wrote: >>> >>>> Hi, >>>> >>>> I am having another problem over here. When I run "ftdm dump 1 5" from >>>> fs_cli I get my channel status as SUSPENDED. This happens for channel >>>> number 5 to 30. Also when I try to make outgoing calls more than 4 in >>>> number, four calls originate properly whereas for the rest it shows >>>> NORMAL_CIRCUIT_CONGESTION. Here is the output: >>>> >>>> span_id: 1 >>>> chan_id: 5 >>>> physical_span_id: 1 >>>> physical_chan_id: 5 >>>> physical_status: ok >>>> physical_status_red: 0 >>>> physical_status_yellow: 0 >>>> physical_status_rai: 0 >>>> physical_status_blue: 0 >>>> physical_status_ais: 0 >>>> physical_status_general: 0 >>>> signaling_status: UP >>>> type: B >>>> state: SUSPENDED >>>> last_state: RESTART >>>> txgain: 0.00 >>>> rxgain: 0.00 >>>> cid_date: >>>> cid_name: >>>> cid_num: >>>> ani: >>>> aniII: >>>> dnis: >>>> rdnis: >>>> cause: NORMAL_UNSPECIFIED >>>> session: (none) >>>> >>>> -- States -- -- Function -- -- >>>> Location -- -- Time Offset -- >>>> DOWN => RESTART [on_dchan_up] >>>> [ftmod_libpri.c:2372] 0ms >>>> RESTART => SUSPENDED [on_timeout_t316] >>>> [ftmod_libpri.c:1952] 120008ms >>>> >>>> Time since last state change: 324790932ms >>>> >>>> Please help me getting out of this. >>>> >>>> Thanks. >>>> >>>> --Ashish >>>> >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >> >

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PRI channels suspended
by Akash Deep Verma 16 Apr '13

16 Apr '13

Hi Ashish, This is a hack but you get what you need. This is because, whenever retry attempt is made 1 channel per pri get started, as there are 30 channels and retry attempts are 3 so the channels that freeswitch load is in between 3-6, so I increased retry attempt and complied it again. As I got rid of this error but it takes some time to load the pri channels (30*10 = 300seconds(5 seconds) max). Also, remember if you dial a call in between restarting phase you may face some error. To avoid this, dial call only after all pri channel get started. Here are the steps: In freeswitch source: Edit libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.h 1. Decrease timeout of channels. #define T316_TIMEOUT_MS_DEFAULT 30000 /* 30 sec */ change this to #define T316_TIMEOUT_MS_DEFAULT 10000 /* 10 sec */ 2. Increase attempts to 30. #define T316_ATTEMPT_LIMIT_DEFAULT 3 change this to #define T316_ATTEMPT_LIMIT_DEFAULT 30 3. Increase maximum limit as according to default(set more than that of default) #define T316_ATTEMPT_LIMIT_MAX 10 change this to #define T316_ATTEMPT_LIMIT_MAX 35 Thanks & Regards, - Akash Deep Verma http://in.linkedin.com/in/akashdeep1

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Newbie question about inbound calls/NAT (yes I checked wiki first:-) )
by Jeff Bernhardt 16 Apr '13

16 Apr '13

Setting up my first test box with Freeswitch. I'm really excited about getting it up and running. I'm trying to get inbound calls working from external using a Vitelity trunk. Let's say my username from Vitelity is "abcd_efg" and password is "password," external IP is 1.2.3.4 and internal is 192.168.10.32. I get registered to Vitelity fine (their web panel shows I'm registered and fs_cli confirms). However, when I make a call inbound, I get: [WARNING] sofia_reg.c:2621 Can't find user [abcd_1.2.3.4(a)192.168.10.32] from <vitelity-inbound-trunk-ip> You must define a domain called '192.168.10.32' in your directory and add a user with the id="abcd_1.2.3.4" attribute and you must configure your device to use the proper domain in it's authentication credentials. I don't know if this is related, but to get around NAT, I have this in vars.xml: <X-PRE-PROCESS cmd="set" data="external_rtp_ip=1.2.3.4"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=1.2.3.4"/> and this in both internal.xml and external.xml: <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> I have a feeling I've misunderstood something fundamental about setting up Freeswitch for inbound access. I've put in the above configs as per what I've found in the wiki, but I'm not sure how to do what the warning says or if it really means I have something not setup right. What am I doing wrong? Thank you.

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IVR, bridge, ring back problem
by andpe 16 Apr '13

16 Apr '13

Welcome. I have a problem with the IVR in FreeSwitch and function bridge. How it works: customer calls IVR, IVR announses customer chooses button 1 number 1 uses function bridge to connect to, say, 1001. during the bridge establishes a connection user (calling party) can not hear the ring signal (silent in handset). when dialed user pick up handset then all works ok (we can talk). Also when dialed user reject the connection I hear busy signal. I have seen in logs Ring Ready not send to calling party. When user calls directly 1001 all works fine. Another question: How to pass music to user when hi waiting for connection when I use bridge function in IVR? I was looking for a solution but I failed to find any. I am new in FreeSwitch. Sory for my english. Regards Andy

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FS Wiki site is down
by Muhammad Shahzad 16 Apr '13

16 Apr '13

It seems FS Wiki site is down at the moment. I am getting this on error page, Can't contact the database server: Unknown MySQL server host ' db.freeswitch.org' (2) (db.freeswitch.org) Thank you. -- Mit freundlichen Grüßen Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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how Boost fs service startup
by shayne.alone＠gmail.com 16 Apr '13

16 Apr '13

Hi, how can I find the reason cause freeswitch startup service, to talk long too mutch? and how can I help it's startup process to go faster? thanks

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PAP2T INCOMPATIBLE_DESTINATION
by Juan Antonio Ibañez Santorum 16 Apr '13

16 Apr '13

Hello, I've got two different PAT2T boxes pointing to my FS box. First one works ok while second what gets INCOMPATIBLE_DESTINATION. Looking at INVITE request I am not be able to get the reason. If you see http://pastebin.com/3emQDfER you will get 2 INVITE, first one works while second no. Do you know why? Regards

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Codec negotiation bug?
by Juan Antonio Ibañez Santorum 16 Apr '13

16 Apr '13

Hello, I am doing some codec negotiation (early) tests but FS does not work as I expected. I can read at wiki: "When FS calls leg B, the list of codecs in outbound-codec-prefs for the SIP profile is reorganized by pushing the codec negotiated above for leg A at the top. If B does not accept any of the codecs, the calls fails, obviously" Codec prefs setted as: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/> FS select PCMA for A leg FS offers following SDP to B leg: v=0 o=FreeSWITCH 1365781615 1365781616 IN IP4 176.31.117.49 s=FreeSWITCH c=IN IP4 176.31.117.49 t=0 0 m=audio 19780 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 m=audio 19780 RTP/AVP 18 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Session progress from B leg contains: v=0 o=root 3311 3311 IN IP4 91.192.110.234 s=session c=IN IP4 91.192.110.234 t=0 0 m=audio 16194 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv FS codec comparison debug look as: sofia_glue.c:5094 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:30:64000] sofia_glue.c:5094 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] sofia_glue.c:3077 Set Codec sofia/external/0034950004588 G729/8000 20 ms 160 samples 8000 bits FS start comparing G729 instead PCMA selected in A leg Is that ok? is there any way to get selected PCMA for B leg without using late negotiation nor disabling transcoding Regards

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Communication between sip soft phones(X-Lite) via freeswitch
by Navnath Sonavne 16 Apr '13

16 Apr '13

Hi All, I have successfully compiled and installed freeswitch on my Linux CentOS. I have registered two softphones of X-Lite in freeswitch with extension numbers are 1100 ,1101. I have installed these softphones on two different machines of windows OS. I have created two XML dial plans for above extension numbers in usr/local/freeswitch/conf/directory/default. Now when i call 1101 from 1100 extension it did not establish a call. I know something is going wrong but what is it? I am not able to figure out. I request you all please help me to establish a call between 1100 and 1101. FYI i am reading Freeswitch 1.0.6 by Anthony Minessale , Michael S. Collins , Darren Schreiber . I am attaching 1100.xml and 1101.xml files. Regards, Navnath.

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Communication between sip soft phones(X-Lite) via freeswitch
by Navnath Sonavne 15 Apr '13

15 Apr '13

Hi, Your response : Probably your users might not be in the default context. To make call from one sip user to another both must be in same context. For testing purpose you may want to keep both of them in default context. In the user_name.xml file put context = default , this may solve your problem. My both extensions are in default context .To crosscheck it i used list_users command at fs_cli console and it shown that both phones are in default context.Here is output of list_users 1100|default|192.168.8.41|default|sofia/internal/sip:1100@192.168.9.165:29072;rinstance=ee8af9a648b2e70d||Vivekanandm|1100 1101|default|192.168.8.41|default|sofia/internal/sip:1101@192.168.8.103:26196;rinstance=a6ab3905d800b8f7||navnath|1101 As you suggested put context = default in user_name.xml but i did not find user_name.xml file in my freeswitch directory. Can you please tell me where to locate this file? Regards, Navnath.

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