Re: [asterisk-users] Digest Username/auth name mismatch
Hi I add the details. The error seems that UAC set different username of digest. But UAC cannot send same username of digest and from for specification. So I want to know how to solve with Asterisk. Register From: sip:a...@192.168.0.1;tag=644056924 To: sip:a...@192.168.0.1 Call-ID: 2457796...@192.168.0.2 CSeq: 125 REGISTER Contact: sip:a...@192.168.0.2:5060 Authorization: Digest username=a...@192.168.0.1, realm=asterisk, nonce=3e635209, uri=sip:192.168.0.1, response=ec89ab3c90316e05d83774630488c61a, algorithm=MD5 Max-Forwards: 70 Expires: 3600 thanks Date: Wed, 1 Sep 2010 15:15:36 +1200 From: li...@venturevoip.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digest Username/auth name mismatch On 30/08/10 2:48 PM, kawanobe tomohito wrote: Hi I want to know how to solve below an error case. Uac cant's change username of from and digest header. I tried to put a...@192.168.0.1 on username of sip.conf.but same error returned. You don't need to have the @192.168.0.1 in there - just make sure the username and password are correct in the user's device. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny
Hi guys, Interesting discussion - I learnt quite a bit. Thanks. That said, no one's yet answered my two original questions. Anyone know? To repeat: 1. When I used the line dateformat=%F %T in the general section of logger.conf, the format in /var/log/asterisk/full did change, but the round brackets around the date remained; the fail2ban/asterisk instructions I was following indicated that they should disappear when I do this. Is this an asterisk version issue (I'm using 1.4.21 - would 1.6 behave in the way described?), a Debian issue (seems unlikely), or something else? Is there some other way to get the round brackets to disappear? This is necessary to get fail2ban to read that file; otherwise, I'll have to log all asterisk NOTICEs through syslog. 2. With alwaysauthreject=yes and using deny= and permit= in sip.conf, attempts from denied IP addresses to register an extension are responded to (denying them, obviously), but not logged as a NOTICE (or anything else, as far as I can tell). Is there a way to enable this logging - or, alternatively, to get asterisk to simply ignore these requests rather than responding? I fully appreciate the frustration others feel re ending up paying for other people's attacks as part of their download limit, or even maxing out because of this. I'm lucky in that I haven't reached that volume yet - I'm with Zen Internet, and have a 50GB monthly limit, which I'm not using anywhere near all of. (that said, I did use 30GB last month, and suspect that the lion's share of that was from attacks, SIP reg or otherwise; this could certainly be an issue in the future.) Instead, my issue was with maxing out my *upload* bandwidth limit (currently only 448kbps), and hence having my whole connection screeching to a halt, with massive packet loss to other applications. At that point, not even (a sane amount of) money helps, as you can't buy a higher upload rate (aside from regrading to ADSL2+, which I'm looking into now). Thanks in advance, Nikhil. -- _ -- 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 routing to SoftSwitch
Any Idea?? On Mon, Aug 30, 2010 at 11:11 AM, Pratik Shrestha pratik...@gmail.comwrote: Oh so sorry. Yes you are right, the 'callee'. We have one soft switch somewhere located in US. When the call comes, then asterisk has to see the callee number in the sip extensions. If the number is not in its extensions, then that call should be routed to that soft switch in US. But the condition is, a number 4237 should be added in callee's number. For example, The call comes destined to '123456', the asterisk will see in its sip extensions. If the number is there then the conversation will start right away. But if the extension is not there, then asterisk with route this call to softswitch but 4237 added, that is callee's number will be 4237123456. I hope you understand me. Thanks a lot. Regards, Pratik On Mon, Aug 30, 2010 at 10:58 AM, Jose P. Espinal j...@slackware-es.comwrote: Hello Pratik, Could you please elaborate your question a little more (describe better your scenario)? Are you sure to be using properly the terms 'caller', and not instead of 'callee'? (caller : makes call, callee : receives call) Regards, Pratik Shrestha wrote: Dear All, First, I am not so much experienced in Asterisk. I need asterisk to route the call to soft switch when the caller is not in its extensions list. And also when routing to soft switch, a number 4327 has to be added in the caller's number and then routed. I think its not so hard in asterisk. Please help me. Regards, Pratik -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs -- _ -- 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] Asterisk routing to SoftSwitch
On 1 Sep 2010, at 10:30, Pratik Shrestha wrote:
Any Idea??
Read
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
But I'm guessing you knew that and are just after getting someone else to do
the work...
Just create a catch-all pattern to match anything your specific dialplan
doesn't (what it does match is completely unknown as you have yet to
satisfactorily describe any of your system to us). I'd guess as _X. but i have
no idea of your setup...
Then to add the prefix just make it dial SIP/softswitchpeer/4327${EXTEN}
EXTEN will be filled with the existing number.
S
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Re: [asterisk-users] Logging the CID from the Privacy Manager
Quoting Matt Riddell li...@venturevoip.com:
Maybe you could do:
Set(CDR(userfield)=${CALLERID(num)})
Before dialing SIP/1000
That looks so simple -- and it actually works! -- although exactly not
in the way that I was expecting. Instead of replacing the contents of
one of the existing fields, a new field, userfield, appeared at the
end of the record containing the number submitted by the caller.
I did try to use the same method to change one of the existing fields,
e.g. src, like this:
Set(CDR(src)=${CALLERID(num)})
But, then I received this error:
[Sep 1 12:26:15] ERROR[12562]: cdr.c:303 ast_cdr_setvar:
Attempt to set the 'src' read-only variable!.
That doesn't seem to be possible. So, I'm happy with your solution.
Thanks, Matt!
Cheers,
Jaap
This message was sent using IMP, the Internet Messaging Program.
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[asterisk-users] Freepbx + Asterisk problem - NEED HELP
Hello, After installing on Ubuntu 10.04 using the tutorial on http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html I have a running instance of Asterisk. PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 4 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 The problem is about asterisk CLI results: asterisk*CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) asterisk*CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefault default In Service --- INFO: related lspci result: 07:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface related dahdi_hardware result: pci::07:00.0 wctdm+ e159:0001 Wildcard TDM400P REV I One more thing: r...@asterisk:~# lsmod|grep dahdi dahdi_echocan_mg2 5729 4 dahdi_transcode 6836 1 wctc4xxp dahdi_voicebus 41854 2 wctdm24xxp,wcte12xp dahdi 210885 12 dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 1675 2 dahdi,hisax it seems my dahdi/system.conf is ok. # Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) fxsks=1 echocanceller=mg2,1 fxsks=2 echocanceller=mg2,2 fxoks=3 echocanceller=mg2,3 fxoks=4 echocanceller=mg2,4 # Global data loadzone= tr defaultzone= tr And this is zapata.conf: [channels] language=en ; include zap extensions defined in AMP #include zapata_additional.conf ; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c - Button %n context=from-pstn faxdetect=incoming echotraining=800 group=0 busydetect=yes busycount=4 hanguponpolarityswitch relaxdtmf=yes callprogress=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=2.0 txgain=2.0 immediate=yes signalling=fxs_ks channel=1-2 Is it anything related to COUNTRY? Any advice? -- _ -- 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] Freepbx + Asterisk problem - NEED HELP
Hi there, Your problem is with the Tiger ISDN kernel module claiming the digium card. I have the following in /etc/modprobe.d/blacklist.conf blacklist hisax blacklist netjet blacklist isdn blacklist mISDN_core blacklist mISDN_ipac Once the netjet driver isn't claiming the card, the dahdi module will work as advertised. aF On 01/09/2010, at 9:58 PM, Mehmet Kuzulugil wrote: Hello, After installing on Ubuntu 10.04 using the tutorial on http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html I have a running instance of Asterisk. PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 4 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 The problem is about asterisk CLI results: asterisk*CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) asterisk*CLI dahdi show channels Chan Extension Context Language MOH InterpretBlocked State pseudodefaultdefault In Service --- INFO: related lspci result: 07:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface related dahdi_hardware result: pci::07:00.0 wctdm+ e159:0001 Wildcard TDM400P REV I One more thing: r...@asterisk:~# lsmod|grep dahdi dahdi_echocan_mg2 5729 4 dahdi_transcode 6836 1 wctc4xxp dahdi_voicebus 41854 2 wctdm24xxp,wcte12xp dahdi 210885 12 dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 1675 2 dahdi,hisax it seems my dahdi/system.conf is ok. # Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) fxsks=1 echocanceller=mg2,1 fxsks=2 echocanceller=mg2,2 fxoks=3 echocanceller=mg2,3 fxoks=4 echocanceller=mg2,4 # Global data loadzone= tr defaultzone= tr And this is zapata.conf: [channels] language=en ; include zap extensions defined in AMP #include zapata_additional.conf ; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c - Button %n context=from-pstn faxdetect=incoming echotraining=800 group=0 busydetect=yes busycount=4 hanguponpolarityswitch relaxdtmf=yes callprogress=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=2.0 txgain=2.0 immediate=yes signalling=fxs_ks channel=1-2 Is it anything related to COUNTRY? Any advice? -- _ -- 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] Freepbx + Asterisk problem - NEED HELP
On Wed, Sep 01, 2010 at 02:58:59PM +0300, Mehmet Kuzulugil wrote: Hello, After installing on Ubuntu 10.04 using the tutorial on http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html I have a running instance of Asterisk. PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 How nice. Why don't you use the latest? 2.3.0[.1]? (2.4.0 should be released shortly, but that's not really something users are expected to guess). DAHDI Version: 2.2.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 4 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 The problem is about asterisk CLI results: asterisk*CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) asterisk*CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefaultdefault In Service What's the output of lsdahdi ? This suggests a misconfigured /etc/dahdi/chan_dahdi.conf or one of its --- INFO: related lspci result: 07:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface related dahdi_hardware result: pci::07:00.0 wctdm+ e159:0001 Wildcard TDM400P REV I One more thing: r...@asterisk:~# lsmod|grep dahdi dahdi_echocan_mg2 5729 4 dahdi_transcode 6836 1 wctc4xxp dahdi_voicebus 41854 2 wctdm24xxp,wcte12xp dahdi 210885 12 dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 1675 2 dahdi,hisax Haveing just 'wctdm' in /etc/dahdi/modules would save you a second or two on loading (and a bunch of log messages). it seems my dahdi/system.conf is ok. # Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) fxsks=1 echocanceller=mg2,1 fxsks=2 echocanceller=mg2,2 fxoks=3 echocanceller=mg2,3 fxoks=4 echocanceller=mg2,4 # Global data loadzone= tr defaultzone= tr And this is zapata.conf: zapata.conf is ignored for asterisk = 1.6.0 . Please tell me you don't use 1.4.x . [channels] language=en ; include zap extensions defined in AMP #include zapata_additional.conf ; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c - Button %n context=from-pstn faxdetect=incoming echotraining=800 group=0 busydetect=yes busycount=4 hanguponpolarityswitch relaxdtmf=yes callprogress=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=2.0 txgain=2.0 immediate=yes signalling=fxs_ks channel=1-2 -- 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
[asterisk-users] 3Com 3102 Phones
Has any advancement been made to get 3102 operational in either a SIP or H323 asterisk environment. A post back in time mentioned a downloader service. From the posts and articles I have read, the NCP is acting like a bootp and tftp server which uploads the configuration to the phone?? Am I close? if so, where does one get the SIP image for he 3102 and 2102 phones? I had 8 donated, but they are useless without a NBX or NCP ? Any specs on how to configure linux to act like one? Thanks in advance -- _ -- 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] ChanSpy getting piled up
Hi list,
Im using asterisk 1.6.0.10 and have following dialplan for doing chanspy
[app-chanspy]
include = app-chanspy-custom
exten = 555,1,Read(SPYNUM,extension)
exten = 555,2,ChanSpy(SIP/${SPYNUM},q)
exten = 555,n,Hangup
but if the channel is hang up or even destroyed the chanspy is not
getting killed.
asteriskcore show channels verbose
.
.
.
SIP/1009-b6c5b398from-internal555 2 Up
ChanSpy SIP/1002,q1009
554:53:1 (None)
SIP/1009-b5004908from-internal555 2 Up
ChanSpy SIP/1002,q1009
-571:-19 (None)
SIP/1009-b50a4e30from-internal555 2 Up
ChanSpy SIP/1002,q1009
-571:-9: (None)
SIP/1009-b50702a8from-internal555 2 Up
ChanSpy SIP/1002,q1009
-571:-5: (None)
SIP/1009-09bafcd0from-internal555 2 Up
ChanSpy SIP/1002,q1009
-570:-57 (None)
.
.
.
Is there a way to cleanup this ?
Regards
Rushikesh
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Re: [asterisk-users] ChanSpy getting piled up
chanspy as best I can tell from the code will not lock on a single device and
when that device goes away exit. What is passed to chanspy is a template for a
channel name. I submitted a patch to add option s so that chanspy would stop
when the one channel I wanted to watch went away or I used * to stop.
https://issues.asterisk.org/view.php?id=14594
--
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http://www.cfmc.com/
On Sep 1, 2010, at 6:24 AM, Rushikesh wrote:
Hi list,
Im using asterisk 1.6.0.10 and have following dialplan for doing chanspy
[app-chanspy]
include = app-chanspy-custom
exten = 555,1,Read(SPYNUM,extension)
exten = 555,2,ChanSpy(SIP/${SPYNUM},q)
exten = 555,n,Hangup
but if the channel is hang up or even destroyed the chanspy is not
getting killed.
asteriskcore show channels verbose
.
.
.
SIP/1009-b6c5b398from-internal555 2 Up
ChanSpy SIP/1002,q1009
554:53:1 (None)
SIP/1009-b5004908from-internal555 2 Up
ChanSpy SIP/1002,q1009
-571:-19 (None)
SIP/1009-b50a4e30from-internal555 2 Up
ChanSpy SIP/1002,q1009
-571:-9: (None)
SIP/1009-b50702a8from-internal555 2 Up
ChanSpy SIP/1002,q1009
-571:-5: (None)
SIP/1009-09bafcd0from-internal555 2 Up
ChanSpy SIP/1002,q1009
-570:-57 (None)
.
.
.
Is there a way to cleanup this ?
Regards
Rushikesh
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Re: [asterisk-users] 3Com 3102 Phones
On Wed, Sep 1, 2010 at 6:09 AM, Barry Fawthrop ba...@isscp.com wrote: Has any advancement been made to get 3102 operational in either a SIP or H323 asterisk environment. A post back in time mentioned a downloader service. From the posts and articles I have read, the NCP is acting like a bootp and tftp server which uploads the configuration to the phone?? Am I close? if so, where does one get the SIP image for he 3102 and 2102 phones? I had 8 donated, but they are useless without a NBX or NCP ? Any specs on how to configure linux to act like one? Thanks in advance -- _ -- 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 I did some digging, NCP seems to stand for Network Call Processor. The 3com asterisk guide has a picture of a device that looks like digiums first asterisk appliance. http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf http://support.3com.com/documents/asterisk/3Com_Asterisk_Admin_Guide.pdf http://www.digium.com/en/products/appliance/ Here is a reference to handing out NCP address via dhcp: http://antjedi.homeip.net/wordpress/?p=25 Best bet might be to sniff some packets, it's not clear to me if NCP uses SIP or not -- _ -- 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] ChanSpy getting piled up
On Wednesday 01 September 2010 09:01 PM, Jim Dickenson wrote: chanspy as best I can tell from the code will not lock on a single device and when that device goes away exit. What is passed to chanspy is a template for a channel name. I submitted a patch to add option s so that chanspy would stop when the one channel I wanted to watch went away or I used * to stop. https://issues.asterisk.org/view.php?id=14594 Hi Jim, Thanks for your reply, Im a new user to asterisk and have very basic knowledge of it. by looking at patch I think you are suggesting me to apply the patch to asterisk source code and recompile my asterisk. Actually this is a production system so I'm not sure whether my Boss will allow me to do it ;) . Do you know any other work around for this ? As you said I need to stop chanspy once the channel wen away. Regards, Rushikesh -- _ -- 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] * and mj
Hello all,
Has anyone have magicjack working with their asterisk? I had patched
chan_sip.c with some code that allows asterisk to do the md5 hash that mjmd5
proxy does. * shows that it is registered with magicjack, but incoming calls
are not even hitting my * box and outgoing calls get congestion. Here is my
relevant configs. I did do a ton of google searching, but it all points to
it should work. And I am stuck. I know my existing dial plan internally
works, as it already works with my GV and ipkall. Any links or suggestions
would be greatly appreciated.
Thanks.
A little snip of sip show peers
magicjack/Exx0167.91.177.705070 OK (55 ms)
I get this in the CLI
-- Executing [12486323...@home-sip-int-in:1]
Dial(SIP/jjonesip-0006, SIP/...@magicjack,30,r) in new stack
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Called ...@magicjack
-- Got SIP response 480 Temporarily Unavailable back from 67.91.177.70
-- SIP/magicjack-0007 is circuit-busy
Sip.conf
[general]
useragent=MagicJack/2.0.554f (SJ Labs); incase they look at
the UA of sip client
register =
Exx01:...@proxy1.detroit.talk4free.com:5070/1...@hom
e-sip-int-in
[magicjack]
context=home-sip-int-in
username=Exx01
authuser=Exx01
type=friend
secret=
port=5070
nat=no
insecure=port,invite
;host=67.106.133.198 ; idk was in a howto
host=proxy1.detroit.talk4free.com
;host=67.91.177.70 ; detroit proxy
;host=vms03.detroit.talk4free.com
;host=67.91.177.77 ; vms03.detroitproxy
useragent=MagicJack/2.0.554f (SJ Labs)
fromuser=Exx01
fromdomain=talk4free.com
dtmfmode=rfc2833
;dtmfmode=inband
qualify=2000
canreinvite=no
disallow=all
allow=ulaw
t38pt_udptl = no
And the code I used to patch chan_sip.c
--- old/channels/chan_sip.c 2009-08-13 10:24:40.0 -0700
+++ new/channels/chan_sip.c 2009-08-22 13:47:29.0 -0700
@@ -8535,6 +8535,32 @@
ast_md5_hash(a2_hash, a2);
snprintf(resp, sizeof(resp), %s:%s:%s, a1_hash, usednonce,
a2_hash);
ast_md5_hash(resp_hash, resp);
+
+
+ /* To a Magicjack domain */
+ if (strstr(uri,talk4free.com))
+ {
+ char callid[256];
+ char newnonce[256];
+ char *c;
+ int i;
+ ast_copy_string(callid, p-callid, sizeof(callid));
+ ast_copy_string(newnonce, p-nonce, sizeof(newnonce));
+
+ strcat(newnonce, _);
+ c = newnonce + strlen(newnonce);
+ char hex[2];
+ hex[1] = 0;
+ for (i = 0; i 8; i++) {
+ hex[0] = newnonce[i];
+ int x = strtol(hex, NULL, 16);
+ *c++ = callid[x];
+ }
+ *c++ = 0;
+
+ snprintf(resp,sizeof(resp),%s:%s:%s, a1_hash, newnonce,
a2_hash);
+ ast_md5_hash(resp_hash, resp);
+ }
}
good_response = keys[K_RESP].s
@@ -11658,6 +11684,31 @@
snprintf(resp,sizeof(resp),%s:%s:%s, a1_hash, p-nonce,
a2_hash);
ast_md5_hash(resp_hash, resp);
+ /* To a Magicjack domain */
+ if (strstr(uri,talk4free.com))
+ {
+ char callid[256];
+ char newnonce[256];
+ char *c;
+ int i;
+ ast_copy_string(callid, p-callid, sizeof(callid));
+ ast_copy_string(newnonce, p-nonce, sizeof(newnonce));
+
+ strcat(newnonce, _);
+ c = newnonce + strlen(newnonce);
+ char hex[2];
+ hex[1] = 0;
+ for (i = 0; i 8; i++) {
+ hex[0] = newnonce[i];
+ int x = strtol(hex, NULL, 16);
+ *c++ = callid[x];
+ }
+ *c++ = 0;
+
+ snprintf(resp,sizeof(resp),%s:%s:%s, a1_hash, newnonce,
a2_hash);
+ ast_md5_hash(resp_hash, resp);
+ }
+
/* only include the opaque string if it's set */
if (!ast_strlen_zero(p-opaque)) {
snprintf(opaque, sizeof(opaque), , opaque=\%s\, p-opaque);
--
_
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Re: [asterisk-users] ChanSpy getting piled up
I had the same need which is why I submitted the patch. I think the feature might finally be added to 1.8, it I remember correctly. I am not aware of any other way around this. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Sep 1, 2010, at 9:12 AM, Rushikesh wrote: On Wednesday 01 September 2010 09:01 PM, Jim Dickenson wrote: chanspy as best I can tell from the code will not lock on a single device and when that device goes away exit. What is passed to chanspy is a template for a channel name. I submitted a patch to add option s so that chanspy would stop when the one channel I wanted to watch went away or I used * to stop. https://issues.asterisk.org/view.php?id=14594 Hi Jim, Thanks for your reply, Im a new user to asterisk and have very basic knowledge of it. by looking at patch I think you are suggesting me to apply the patch to asterisk source code and recompile my asterisk. Actually this is a production system so I'm not sure whether my Boss will allow me to do it ;) . Do you know any other work around for this ? As you said I need to stop chanspy once the channel wen away. Regards, Rushikesh -- _ -- 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] * and mj
Looks to me like the problem is with your dial command; You're trying to do dial(sip/a,30,r) Instead of Dial(sip/a/b,30,r) In simple terms, if I have extension sip/170 and I do Dial(sip/170,30,r) That's ok because sip/170 is an extension But the magicjack is a trunk, so I have to do Dial(sip/170/w5551212,30,r) -- _ -- 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] * and mj
Jeff Jones jeff.jjo...@gmail.com wrote: Has anyone have magicjack working with their asterisk? I had patched chan_sip.c I highly recommend running 'mjproxy' on a host with a public IP instead of patching asterisk; I can confirm that it is a good solution and asterisk can be configured as a trunk for incoming and outgoing calls. -- _ -- 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] help with dialplan
On Tue, Aug 31, 2010 at 9:34 AM, Danny Nicholas da...@debsinc.com wrote: Why not just copy the _1NXXNXX line into the remote context? Well, that could be done, and probably would be a good tactic if you have lots of DID's and want to do db lookup or something to direct the next call leg. But, if you only have one or two DID's, all the machinery and programming seem a bit overkill. murf -- -- Steve Murphy ParseTree Corp -- _ -- 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] MOH in the middle of the call
Hi, I have a very strange problem. In the middle of the call the MOH starts for 30 seconds approximately. After this the call run normally. Anybody have an ideia or has some similar problem? Thanks in advance!! -- Atenciosamente, --- Dario Quiroz Analista de Suporte (71) 9275-9080 gtalk: darioqui...@gmail.com --- -- _ -- 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] MOH in the middle of the call
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dario Quiroz Subject: [asterisk-users] MOH in the middle of the call Hi, I have a very strange problem. In the middle of the call the MOH starts for 30 seconds approximately. After this the call run normally. Anybody have an ideia or has some similar problem? Thanks in advance!! You haven't provided enough information. Guesses would be that it is a normal thing or that you are getting some kind of perhaps SIP error that is causing a momentary disconnect, triggering MOH until the condition resolves itself. -- _ -- 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] libpri 1.4.11.4 Now Available
The Asterisk Development Team has announced the release of libpri 1.4.11.4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ The release of libpri 1.4.11.4 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are some of the issues resolved in this release: * Fix issue where calling name is not successfully processed on inbound QSIG PRI calls from Mitel PBX. (Closes issue #17619. Reported by: jims8650. Patched by rmudgett) * Added missing code specified by Q.921 (Figure B.8 Page 85) when receive RNR in Timer Recovery state. (Closes issue #16791. Reported by: alecdavis. Patched by alecdavis) * Fixed issue where incoming calls specifying the channel using a slot map could not negotiate a B channel correctly. * Add support to receive ECMA-164 2nd edition OID name ROSE messages. * Fixed issue where ISDN BRI PTMP TE does not recover from line faults. (Closes issue #17570. Reported by: jcovert. Patched by rmudgett) * Q.921 improvements from comparing Q.921 SDL diagrams with implementation. * Q.921/Q.931 message debug output improvements. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.11.4 Thank you for your continued support of Asterisk! -- _ -- 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] ITSP with DDIs (or DIDs) from India
Anyone know of an ITSP that can offer DDIs (or DIDs) from India? -- _ -- 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] MOH in the middle of the call
Danny Nicholas schrieb: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dario Quiroz *Subject:* [asterisk-users] MOH in the middle of the call Hi, I have a very strange problem. In the middle of the call the MOH starts for 30 seconds approximately. After this the call run normally. Anybody have an ideia or has some similar problem? Thanks in advance!! You haven’t provided enough information. Guesses would be that it is a normal thing or that you are getting some kind of perhaps SIP error that is causing a momentary disconnect, triggering MOH until the condition resolves itself. i had some problems like this, but only when a snom phone transfered a call. if you use asterisk 1.6.x this could also be an answer bug which is allready been fixed. this bug cause some strange issues with moh and wrong codec write formats. but without further information its just a random guess ;) best regards steve smith -- _ -- 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 routing to SoftSwitch
Thanks Steve..
On Wed, Sep 1, 2010 at 5:13 PM, Steve Howes steve-li...@geekinter.netwrote:
On 1 Sep 2010, at 10:30, Pratik Shrestha wrote:
Any Idea??
Read
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
But I'm guessing you knew that and are just after getting someone else to
do the work...
Just create a catch-all pattern to match anything your specific dialplan
doesn't (what it does match is completely unknown as you have yet to
satisfactorily describe any of your system to us). I'd guess as _X. but i
have no idea of your setup...
Then to add the prefix just make it dial SIP/softswitchpeer/4327${EXTEN}
EXTEN will be filled with the existing number.
S
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[asterisk-users] NCS - Cablemodem
Hi all, I am configuring asterisk in a cable modem network, using a motorola TM401A. I can make calls from the MTA but I can receive, display the following error: -- Executing [1...@alberti:1] Dial(OSS/dsp, MGCP/aaln/1...@0-13-11-82-bd-a.ssw.intercal.net|30) in new stack [Sep 2 00:10:53] NOTICE[28062]: chan_mgcp.c:3572 mgcp_request: Asked to get a channel of unsupported format '0' [Sep 2 00:10:53] WARNING[28062]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'MGCP' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) extensions.conf exten = 1500,1,Dial(MGCP/aaln/1...@0-13-11-82-bd-a.ssw.dominio.net,30) exten = 1500,n,Hangup() mgcp.conf [general] port=2427 bindaddr=0.0.0.0 disallow=all allow=alaw language=es [0-13-11-82-bd-a.ssw.dominio.net] context=internos host=10.30.15.254 ncs=1 nat=no slowsequence=yes wcardep=aaln/* callwaiting=no callreturn=yes cancallforward=yes canreinvite=no threewaycalling=no immediate=no transfer=no dtmfmode=inband callerid = 1500 1500 line = aaln/1 ssw*CLI core show version Asterisk 1.4.17 built by root @ ssw on a x86_64 running Linux on 2010-04-14 13:45:12 UTC ssw*CLI mgcp show endpoints Gateway '0-13-11-82-bd-a.ssw.dominio.net' at 10.30.15.254 (Static) -- 'aaln/1...@0-13-11-82-bd-a.ssw.dominio.net in 'alberti' is idle ssw*CLI mgcp audit endpoint aaln/1...@0-13-11-82-bd-a.ssw.dominio.net Posting Request: AUEP 3 aaln/1...@0-13-11-82-bd-a.ssw.dominio.net MGCP 1.0 NCS 1.0 F: A to 10.30.15.254:2427 MGCP read: 200 3 OK A: a:PCMU;PCMA;G728;G729;G729E;G726-16;G726-24;G726-32;G726-40, p:10-30, b:19-100, e:on, t:1, s:off, v:L;fxr;rg;xal;x-xl;fm;lcs;sst;x-jc;x-pol;xrm, m:sendrecv;sendonly;recvonly;inactive;netwloop;netwtest;replcate;confrnce, dq-gi, sc-rtcp: 81/70;81/71;82/70;82/71;80/70;80/71, sc-rtp: 62/51;62/50;64/51;64/50;60/51;60/50 A: a:telephone-event, fmtp:telephone-event 0-15,144,149,159 A: a:image/t38, p:10-30, b:25-64, dq-gi from 10.30.15.254:2427 Verb: '200', Identifier: '3', Endpoint: 'OK', Version: '(null)' 4 headers, 0 lines Thanks, -- _ -- 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
