[asterisk-users] t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED
Hi everyone. We have a problem here... Hope somebody can give us some hints. We have a HP ProLiant DL180 G6 Server with a Debian/Lenny sytem. Asterisk 1.4.21.2 (1.4.21.2~dfsg-3+lenny1) with zaptel (1.4.11) and libpri (1.4.3) is installed. There is a QuadBRI-Card installed: # lspci -vv -s 06:04.0 06:04.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Subsystem: Cologne Chip Designs GmbH Device b752 Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- INTx- Interrupt: pin A routed to IRQ 30 Region 0: I/O ports at cc00 [size=8] Region 1: Memory at fb6ff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- zttest gives me an average of 99.992% and zttool shows no alarms. But every about 3,5 minutes we get this (with debug span 1 enababled): 1 -- Timeout occured, restarting PRI 1 q921.c:859 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED 1 Sending Set Asynchronous Balanced Mode Extended 1 q921.c:534 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH == Primary D-Channel on span 1 down [Apr 10 12:16:05] WARNING[28541]: chan_zap.c:2498 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! 1 Sending Set Asynchronous Balanced Mode Extended 1 -- Got UA from network peer Link up. 1 -- Restarting T203 counter == Primary D-Channel on span 1 up % cat /etc/zaptel.con # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: ztqoz/1/1 quadBRI PCI ISDN Card 1 Span 1 [TE] (cardID 0) (MASTER) span=1,1,3,ccs,ami # termtype: te bchan=1-2 dchan=3 # Span 2: ztqoz/1/2 quadBRI PCI ISDN Card 1 Span 2 [TE] (cardID 0) span=2,2,0,ccs,ami # termtype: te bchan=4-5 dchan=6 # Span 3: ztqoz/1/3 quadBRI PCI ISDN Card 1 Span 3 [TE] (cardID 0) span=3,3,0,ccs,ami # termtype: te bchan=7-8 dchan=9 # Span 4: ztqoz/1/4 quadBRI PCI ISDN Card 1 Span 4 [TE] (cardID 0) span=4,4,0,ccs,ami # termtype: te bchan=10-11 dchan=12 # Global data loadzone= at defaultzone = at % cat /etc/asterisk/zapata.conf [channels] language=de switchtype=euroisdn pridialplan=unknown prilocaldialplan=dynamic priindication=passthrough context=incoming immediate=no usecallingpres=yes usecallerid=yes group=1 nationalprefix=00 internationalprefix=000 signalling=bri_cpe echocancel=Yes overlapdial=Yes ; group=2 ; signalling=bri_cpe ; context=incoming ; channel = 10-11 ; channel = 1-2 ; channel = 4-5 ; channel = 7-8 ; channel = 10-11 (Only one span is connected to ISDN right now.) qozap is loaded and ztcfg -v gives me: Zaptel Version: 1.4.11 Echo Canceller: MG2 Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) 12 channels to configure. Any idea what this could mean and how this could be fixed? Any help would be helpful. Thx. Greetings, - Darsha -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please sign Petition - Stop Child Labour
On Fri, Apr 09, 2010 at 11:27:34AM -0400, Martin wrote: Are you sure writing to the right list??? Thanks for helping that mail defeat my spam filter. Please don't help spam by quoting it. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED
Hi! On 04/10/2010 02:04 PM, Tzafrir Cohen wrote: On Sat, Apr 10, 2010 at 12:32:49PM +0200, Darshaka Pathirana wrote: We have a problem here... Hope somebody can give us some hints. We have a HP ProLiant DL180 G6 Server with a Debian/Lenny sytem. Asterisk 1.4.21.2 (1.4.21.2~dfsg-3+lenny1) with zaptel (1.4.11) and libpri (1.4.3) is installed. Sadly those packages diverge from the mainline Asterisk in one important aspect: they use the bristuff patch in both asterisk and libpri. Hmm. Yes I know. Why sadly? Anything bad about it? We need some features which are missing without bristuff. I also thought qozap needs a bristuffed Asterisk/libpri... Do you need NT PtMP support (It doesn't look that way from your system)? If not, I wonder if you would consider a backport of latest Squeeze packages, which I maintain: http://updates.xorcom.com/pkg-voip/ Indeed this setup just needs PtP. Is PtMP not supported in Asterisk 1.6? Greetings, - Darsha -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Repeated: Got SIP response 489 Bad event back from
Hi All, I've two asterisk servers on the same LAN, both 1.4, and I keep getting Got SIP response 489 Bad event back from 192.168.3.10 No idea whats causing it. The only references I can find mentions NATing issues, but these are on the same LAN so NAT shouldn't be an issue. 3.10 does authenticate into the server logging the error. The error appears in the log every 1m20s (ish) Any ideas? Thanks, Adrian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Repeated: Got SIP response 489 Bad event back from
On Sat, Apr 10, 2010 at 6:35 AM, Adrian Marsh adrian.ma...@ubiquisys.com wrote: Hi All, I’ve two asterisk servers on the same LAN, both 1.4, and I keep getting “Got SIP response 489 Bad event back from 192.168.3.10” No idea whats causing it. The only references I can find mentions NATing issues, but these are on the same LAN so NAT shouldn’t be an issue. 3.10 does authenticate into the server logging the error. The error appears in the log every 1m20s (ish) Is 3.10 on a SIP trunk to the other asterisk box? Is qualify=yes on this SIP trunk? I think you'll find that if you run an ngrep/tcpdump on port 5060 on the box receiving the error it will send out an OPTIONS or NOTIFY (I can't remember which) and then you'll see the 489 Bad Event. Grab a trace of the SIP traffic and post it, its the only way to know for sure though. -- James Any ideas? Thanks, Adrian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sending RTP media to a different server than SIP Signaling
Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
On Sat, Apr 10, 2010 at 8:50 AM, Tarek Sawah tareksa...@hotmail.com wrote: Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 I don't believe this can be done in asterisk by itself, but you may be able to use the Linux conntrack stuff (http://netfilter.org/) to rewrite the SDP host information... However, if you want to dive into the world of OpenSIPS, I know you can do this with an OpenSIPS/MediaProxy setup between your asterisk box and your provider. -- James _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tones detection
Hi Jerry, On Thu, Apr 8, 2010 at 6:54 PM, Jerry Geis ge...@pagestation.com wrote: I am looking for something in asterisk that will let me record a wav file in asterisk (which I know how to do) then some other command (external or dialplan) that would read the wave file and tell me if a certain tone or frequency is present. Is this in asterisk already - any way to do it? Thanks You might want to look into the PipeWave tools: http://www.cardiff.ac.uk/psych/home2/CullingJ/pipewave.html The tools can generate a FFT (fast-fourier transform) of a wav file which converts the data into the frequency domain, which should allow you to tell if a certain frequency is present. -- James Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + DRBD Performance
Hi, Has anyone had any experience using DRBD to mirror an entire asterisk machine? If so, is there a performance issue at all when people are recording voicemails and the like? It seems like that could generate quite a bit of traffic. Also, do you bother to mirror the log files as well? Thanks. -- James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callerid over IAX Trunks
Don't have a system to test this right now, but read somewhere this was a 2 steps solution: 1) Leave the callerid in your tunk definition blank (in your example the 999 username) 2) Use brakets around the callerid definition of your peers: callerid= 200 (extension 200 in your example) Let us know if it worked. Alyed 2010/4/9 Ye Liu jaux...@gmail.com Hello everyone, I'm fairly new to asterisk and this list. Currently I'm working on IAX trunks to send/receive calls between 2 asterisk boxes with asterisk 1.6.1.1+asterisk gui 2.0. After some work in the gui, two boxes can send/receive calls to/from the other just fine, the only problem I have is the caller id. Here is my setup: 1. on both boxes, I added an IAX user in the gui, say the extension and password are 999 2. I then created IAX trunks for each box using 999 as username and password, hostname/IP was set to be other box's IP 3. when done, from the system status panel, I saw the trunks successfully registered to the other box 4. then I added Outgoing Call Rules to each box: for box1, _2XX -- to_box2_trunk for box2, _1XX -- to_box1_trunk This setup works ok, the only problem is caller id, i.e. when extension(200) from box2 calls to extension(100) from box1, the call can be made but the caller id displayed on 100 is 999 not 200. I have been on this problem for some time already, could anyone here give me a bit help please? -- Ye Liu (AKA @jaux) http://jaux.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI - Native ZAP bridge fails - Is this my patch?
Hi Guys, I am calling out 416-999- on Channel 1 of PRI and then calling 416-999- on Channel 2 of PRI. When the two channels are going to be ZAP native bridged, both channels hangup and CLI show PRI cause (16). Asterisk Verbose *(Channel 1 already connected to party)*: -- Requested transfer capability: 0x00 - SPEECH -- Called g0/416999 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap-bridge, s, 8) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Here is PRI debug, starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2967 q931_release: call 32864 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: RELEASE COMPLETE (90) q931.c:3766 q931_receive: call 32864 on channel 1 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Terminator) Message type: RELEASE (77) q931.c:3801 q931_receive: call 32865 on channel 2 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: RELEASE COMPLETE (90) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null System Info: *Bell Canada PRI* *Asterisk 1.4.21.2 * *Lib PRI 1.4.10* Is this my patch? https://issues.asterisk.org/view.php?id=7494 Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
Just a week ago, I have been in the same situation. Provider was changing from Cisco gateways to I think Nextone and hence provided me many IPs. I found out that the media IPs don't matter and just played around with my NAT settings and all calls can go through just fine by using simply: host=111.111.111.111 and the 111.111.111.111 is just their SIP signaling IP. Their gateway will then transfer asterisk to proper gateways for media. Just give it a try; it should work. But my efforts on finding anything regarding this failed on Google as well. P.S. the voip provider name starts with a T and end with A. Regards, Bruce On Sat, Apr 10, 2010 at 11:50 AM, Tarek Sawah tareksa...@hotmail.comwrote: Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk script to repeat dial of a number
Say, I'm looking for a simple way to dial a number repeatedly for two minutes at a time. The purpose is to busy up a faulty analogue line in an incoming hunt group. Tx Shaun-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk script to repeat dial of a number
LOLthis is just what I was facing 4 days ago. Unfortunately, Asterisk doesn't provide a software feature in Zaptel to do a BUSY. But people on the list suggest that one should call the telephone company and ask them to busy it. If you have the resource and don't mind the bill of calling the bad line with another line (which is still not full proof because someone else could be calling during that time) then check into spool files and do a little bashscript to run in put files in /var/spool/asterisk/outgoing for calls every two minutes. Oh, if you have access to the box, short-circuit the telco line at the telco demarc. http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out-Bruce 2010/4/10 Shaun Wingrin voi...@gmail.com Say, I'm looking for a simple way to dial a number repeatedly for two minutes at a time. The purpose is to busy up a faulty analogue line in an incoming hunt group. Tx Shaun -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
you got the name EXACTLY! i already am doing what you suggest but facing problems with some destinations and they claim that the problem is with my Asterisk server not their routes! -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sat, 10 Apr 2010 15:50:52 -0400 From: bruceb...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling Just a week ago, I have been in the same situation. Provider was changing from Cisco gateways to I think Nextone and hence provided me many IPs. I found out that the media IPs don't matter and just played around with my NAT settings and all calls can go through just fine by using simply: host=111.111.111.111 and the 111.111.111.111 is just their SIP signaling IP. Their gateway will then transfer asterisk to proper gateways for media. Just give it a try; it should work. But my efforts on finding anything regarding this failed on Google as well. P.S. the voip provider name starts with a T and end with A. Regards, Bruce On Sat, Apr 10, 2010 at 11:50 AM, Tarek Sawah wrote: Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Being attacked by an Amazon EC2 ...
Just a heads-up ... my home asterisk server is being flooded by someone from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it - they're trying to send SIP subscribes to one account - and they're flooding the requests in - it's averaging some 600Kbits/sec of incoming UDP data or about 200 a second )-: This is much worse than anything else I've seen. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Being attacked by an Amazon EC2 ...
Its a good idea tos setup Fail2ban, instructions for which are on voip-info.org. It at least blocks such IP addresses, hopefully prompting the attackers to move their attack somewhere else and leave you alone. Another good idea is to lookup in whois database this IP address and see if you can find contact info for the person responsible for this IP address. Then contact them and let them know about this incident. You can also try to ask your ISP if they can block it on their end. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-10 5:39 PM, Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net wrote: Just a heads-up ... my home asterisk server is being flooded by someone from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it - they're trying to send SIP subscribes to one account - and they're flooding the requests in - it's averaging some 600Kbits/sec of incoming UDP data or about 200 a second )-: This is much worse than anything else I've seen. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
Oh, I see. I haven't done a lot of testing on this new IP since the change of gateways happened but I did Canada calls and they go fine. However, this exact provider lies down to their teeth when it comes to problems of call quality and calls not routing. They never accept faults. They even have problems sending calls to Canada and USA. They failed to pass calls to India as well over times. I had a funny issue where they were blocking one specific area code in USA without even telling us. It was just a regular area code. They told me it was blocked but I know it was a lie because they wanted to cover their a$$ as the route was down and it wasn't blocked. I doubt the problem is with sending calls to different media gateway as I think SIP signals take care of that. Just like canreinvite feature. But I reserve the right to be wrong. -Bruce On Sat, Apr 10, 2010 at 4:45 PM, Tarek Sawah tareksa...@hotmail.com wrote: you got the name EXACTLY! i already am doing what you suggest but facing problems with some destinations and they claim that the problem is with my Asterisk server not their routes! -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sat, 10 Apr 2010 15:50:52 -0400 From: bruceb...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling Just a week ago, I have been in the same situation. Provider was changing from Cisco gateways to I think Nextone and hence provided me many IPs. I found out that the media IPs don't matter and just played around with my NAT settings and all calls can go through just fine by using simply: host=111.111.111.111 and the 111.111.111.111 is just their SIP signaling IP. Their gateway will then transfer asterisk to proper gateways for media. Just give it a try; it should work. But my efforts on finding anything regarding this failed on Google as well. P.S. the voip provider name starts with a T and end with A. Regards, Bruce On Sat, Apr 10, 2010 at 11:50 AM, Tarek Sawah wrote: Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How Cisco ATA 186 through SCCP with skinny.conf ?!
Hi people, I have a Cisco ATA 186 which understands only the SCCP protocoll, therefore I am a pure beginner and I hope that anybody of you could help me. How will I configure the ATA which has 2 analog ports? For any support I would kindly thank you Tamer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
we started with them two days ago .. and we are facing plenty of False Answer cases on several destinations although ppl said they have a policy against FAS.. anyway i don't know i will be looking into another method to send the RTP to another server, thanks for the info Date: Sat, 10 Apr 2010 18:06:22 -0400 From: bruceb...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling Oh, I see. I haven't done a lot of testing on this new IP since the change of gateways happened but I did Canada calls and they go fine. However, this exact provider lies down to their teeth when it comes to problems of call quality and calls not routing. They never accept faults. They even have problems sending calls to Canada and USA. They failed to pass calls to India as well over times. I had a funny issue where they were blocking one specific area code in USA without even telling us. It was just a regular area code. They told me it was blocked but I know it was a lie because they wanted to cover their a$$ as the route was down and it wasn't blocked. I doubt the problem is with sending calls to different media gateway as I think SIP signals take care of that. Just like canreinvite feature. But I reserve the right to be wrong. -Bruce On Sat, Apr 10, 2010 at 4:45 PM, Tarek Sawah wrote: you got the name EXACTLY! i already am doing what you suggest but facing problems with some destinations and they claim that the problem is with my Asterisk server not their routes! -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sat, 10 Apr 2010 15:50:52 -0400 From: bruceb...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling Just a week ago, I have been in the same situation. Provider was changing from Cisco gateways to I think Nextone and hence provided me many IPs. I found out that the media IPs don't matter and just played around with my NAT settings and all calls can go through just fine by using simply: host=111.111.111.111 and the 111.111.111.111 is just their SIP signaling IP. Their gateway will then transfer asterisk to proper gateways for media. Just give it a try; it should work. But my efforts on finding anything regarding this failed on Google as well. P.S. the voip provider name starts with a T and end with A. Regards, Bruce On Sat, Apr 10, 2010 at 11:50 AM, Tarek Sawah wrote: Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
- Tarek Sawah tareksa...@hotmail.com wrote: we started with them two days ago .. and we are facing plenty of False Answer cases on several destinations although ppl said they have a policy against FAS.. anyway i don't know i will be looking into another method to send the RTP to another server, The IP address (and port) of where to send audio is negotiated when the call is setup. You can't change it or specify an IP address to use. Even if you did change the IP address you would be sending it to the port associated with the session on the other media gateway. That would just not work. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote registering fails
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm trying to test with a friend who has an Asterisk in his office with the Asterisk which I have in my house. Then I have an extension that he is trying to register remotely. Trying with the Twinkle client, I see that it is registered: - --- 400/400190.0.163.57 D N 5060 OK (35 ms) - --- but to the few seconds I obtain the following thing in Asterisk CLI: - --- 400/400190.0.163.57 D N 5060 UNREACHABLE - --- And Twinkle gives an error 408 request timeout. And when he tries to make the register through his Asterisk instead of use Twinkle, after a little while he obtains errors of this type: - --- [Apr 10 19:07:18] NOTICE[16848]: chan_sip.c:7618 sip_reg_timeout:-- Registration for '4...@myremotehome.com' timed out, trying again (Attempt #138) - --- This is the configuration that I'm using for the extension: - --- [400] username=400 type=friend secret=passwd qualify=yes callerid=Daniel 400 host=dynamic nat=no context=from-internal mailbox=...@voicemail canreinvite=no - --- I tried with both nat=yes ---as it is possible to be observed above--- and nat=no, and we always obtain the same behavior. My Asterisk server is installed in the same firewall with GNU/Linux. I don't believe that it is a problem with the ports since the client registers itself at some time. Whatever happens, I'm allowing connections for the remote IP to the 5060 tcp/UDP port and 1:2 UDP in the firewall. The router that it is ahead has these ports redirected to the firewall. Also I'm using externhost, externip and localnet in /etc/asterisk/sip.conf Which can be the problem? Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvBAFQACgkQZpa/GxTmHTe0mgCcCmDNhkMm3DMc/Ckd7AAzZneF 4ngAn0SL/IC58kNDktcRsxJOaKPoAuCL =Ve4J -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD reporting NOTSURE most of the time
On Tue, Mar 23, 2010 at 9:06 PM, Steve Moran s...@matara.net wrote: I am running Asterisk and using Answer machine detection with call files on a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over 50,000 outbound calls last week, and 70% said NOTSURE). Hi. Did you ever resolve this? I am having the same problem as you when I use AMD with outgoing calls through my Vitelity line. Sending the calls out PSTN seems to work as normal. I tried tweaking the threshold setting as someone else pointed out but it didn't make any difference. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + DRBD Performance
On Sat, Apr 10, 2010 at 9:50 AM, James Lamanna jlama...@gmail.com wrote: Hi, Has anyone had any experience using DRBD to mirror an entire asterisk machine? Entire, no. Specific/Important mounts yes. If so, is there a performance issue at all when people are recording voicemails and the like? I haven't seen any performance issues, but most installs that I have done aren't recording a ton of messages at a time. I don't have any statistics, but if you had more information on the installation size it would be easier to say. I also have MySQL running on a DRBD mirror, and don't have any problems with updates. I also do all of the replication for DRBD on a cluster interface, which does not pass any VoIP traffic. It seems like that could generate quite a bit of traffic. Also, do you bother to mirror the log files as well? I don't mirror the log files. just: /etc/asterisk/ /var/lib/asterisk /var/spool/asterisk and tftp/http/mysql directories if used. If you really want to test it out, create the mirror and do some disk throughput testing. That way you can validate your specific hardware and network infrastructure to what you expect. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] over running my did's
I have a did with 20 channels from didforsale. that we use to let local members call to listen to a conference several times a week without long distance charges. The upcoming call is getting more interest than usual and from places that are not local so we want to use a free conference service in addition to the local conference. How can I setup a conference on my asterisk box for the people that normally call in there and also call an outside number for those that are above and beyond the 20 lines channels I can provide and the are long distance anyways so a number here or a number in iowa doesn't make them any difference. is there a way that I could call the outside conference # and then transfer it to a local asterisk conference and then hang up can call the local asterisk conference back - and if I do that how do I hang up the long distance conference when it is done? I seem to be missing some basic understanding here. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] over running my did's
On 10/04/2010 9:24 PM, Timothy C Litwiller wrote: I have a did with 20 channels from didforsale. that we use to let local members call to listen to a conference several times a week without long distance charges. The upcoming call is getting more interest than usual and from places that are not local so we want to use a free conference service in addition to the local conference. How can I setup a conference on my asterisk box for the people that normally call in there and also call an outside number for those that are above and beyond the 20 lines channels I can provide and the are long distance anyways so a number here or a number in iowa doesn't make them any difference. is there a way that I could call the outside conference # and then transfer it to a local asterisk conference and then hang up can call the local asterisk conference back - and if I do that how do I hang up the long distance conference when it is done? I seem to be missing some basic understanding here. I would call into the free conference service and then transfer that call into my meetme conference. If you're using Trixbox you can use the MeetMe web control to disconnect the call when you're done. You can also disconnect calls from the asterisk cli using the soft hangup command. Darren Wiebe dar...@aleph-com.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote registering fails
Daniel, you are having a problem often seen in pre 1.4.14 versions. Before this release srvlookup=no was the default for sip.conf and guess the same for iax.conf . So if you are working with a previous release just add this parameter .. but change it to serverlookup=yes under your iax.conf [general] section. Alyed 2010/4/10 Daniel Bareiro daniel-lis...@gmx.net -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm trying to test with a friend who has an Asterisk in his office with the Asterisk which I have in my house. Then I have an extension that he is trying to register remotely. Trying with the Twinkle client, I see that it is registered: - --- 400/400190.0.163.57 D N 5060 OK (35 ms) - --- but to the few seconds I obtain the following thing in Asterisk CLI: - --- 400/400190.0.163.57 D N 5060 UNREACHABLE - --- And Twinkle gives an error 408 request timeout. And when he tries to make the register through his Asterisk instead of use Twinkle, after a little while he obtains errors of this type: - --- [Apr 10 19:07:18] NOTICE[16848]: chan_sip.c:7618 sip_reg_timeout:-- Registration for '4...@myremotehome.com' timed out, trying again (Attempt #138) - --- This is the configuration that I'm using for the extension: - --- [400] username=400 type=friend secret=passwd qualify=yes callerid=Daniel 400 host=dynamic nat=no context=from-internal mailbox=...@voicemail canreinvite=no - --- I tried with both nat=yes ---as it is possible to be observed above--- and nat=no, and we always obtain the same behavior. My Asterisk server is installed in the same firewall with GNU/Linux. I don't believe that it is a problem with the ports since the client registers itself at some time. Whatever happens, I'm allowing connections for the remote IP to the 5060 tcp/UDP port and 1:2 UDP in the firewall. The router that it is ahead has these ports redirected to the firewall. Also I'm using externhost, externip and localnet in /etc/asterisk/sip.conf Which can be the problem? Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvBAFQACgkQZpa/GxTmHTe0mgCcCmDNhkMm3DMc/Ckd7AAzZneF 4ngAn0SL/IC58kNDktcRsxJOaKPoAuCL =Ve4J -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote registering fails
Sorry, the parameter should be. srvlookup=yes Alyed 2010/4/10 Alyed al...@vivoxie.com Daniel, you are having a problem often seen in pre 1.4.14 versions. Before this release srvlookup=no was the default for sip.conf and guess the same for iax.conf . So if you are working with a previous release just add this parameter .. but change it to serverlookup=yes under your iax.conf [general] section. Alyed 2010/4/10 Daniel Bareiro daniel-lis...@gmx.net -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm trying to test with a friend who has an Asterisk in his office with the Asterisk which I have in my house. Then I have an extension that he is trying to register remotely. Trying with the Twinkle client, I see that it is registered: - --- 400/400190.0.163.57 D N 5060 OK (35 ms) - --- but to the few seconds I obtain the following thing in Asterisk CLI: - --- 400/400190.0.163.57 D N 5060 UNREACHABLE - --- And Twinkle gives an error 408 request timeout. And when he tries to make the register through his Asterisk instead of use Twinkle, after a little while he obtains errors of this type: - --- [Apr 10 19:07:18] NOTICE[16848]: chan_sip.c:7618 sip_reg_timeout:-- Registration for '4...@myremotehome.com' timed out, trying again (Attempt #138) - --- This is the configuration that I'm using for the extension: - --- [400] username=400 type=friend secret=passwd qualify=yes callerid=Daniel 400 host=dynamic nat=no context=from-internal mailbox=...@voicemail canreinvite=no - --- I tried with both nat=yes ---as it is possible to be observed above--- and nat=no, and we always obtain the same behavior. My Asterisk server is installed in the same firewall with GNU/Linux. I don't believe that it is a problem with the ports since the client registers itself at some time. Whatever happens, I'm allowing connections for the remote IP to the 5060 tcp/UDP port and 1:2 UDP in the firewall. The router that it is ahead has these ports redirected to the firewall. Also I'm using externhost, externip and localnet in /etc/asterisk/sip.conf Which can be the problem? Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvBAFQACgkQZpa/GxTmHTe0mgCcCmDNhkMm3DMc/Ckd7AAzZneF 4ngAn0SL/IC58kNDktcRsxJOaKPoAuCL =Ve4J -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
