Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug
Hi Nothing will stop the behaviour you are seeing. A SIP reload will clear the realtime cache thus stopping the asterisk server knowing where the realtime sip endpoint is until the endpoint re-registers. The question here is, why are you doing SIP reloads? Once you are using RealTime architecture for SIP, sip reloads become unnecessary unless you are making modifications to the general section of your sip.conf and why would you need to do that regularly? Regards Ish On Wed, 2012-02-15 at 12:52 +0530, DHAVAL INDRODIYA wrote: i tried it and it wont work with rtcachefriend=yes On Fri, Feb 10, 2012 at 11:56 PM, JR Richardson jmr.richard...@gmail.com wrote: I am facing an issue with Peer registration in my asterisk server . I am using asterisk version 1.8.5.0 and using SIP real-time architecture.when i am doing registration it registered fine on asterisk as peer is available in Database. But now i am doing 'sip reload' or 'reload' due to some reason my peer registration is going out and i cannot able to call that peer even though in SIP client it shows me 'registered'. Can any body elaborate on this issue which settings i need to put in sip.conf. I also tried to follow this patch https://issues.asterisk.org/view.php?id=14196 But it allready applied in code base so why it wont work? Here is my sip.conf settings. [general] context=from-internal; Default context for incoming cal rtcachefriends=no rtupdate=yes rtautoclear=yes rtsavesysname=yes callcounter = yes callevents=yes bindport=5060; UDP Port to bind to (SIP standard port is 5060) srvlookup=yes; Enable DNS SRV lookups on outbound calls pedantic=yes; Enable slow, pedantic checking for Pingtel tos=184; Set IP QoS to either a keyword or numeric val tos_sip=cs3; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpiry=3600; Max length of incoming registration we allow defaultexpiry=120; Default length of incoming/outoing registration preferred_codec_only=yes disallow=all; First disallow all codecs allow=ulaw; Allow codecs in order of preference allow=alaw insecure=invite language=en ; Default language setting for all users/peers rtpholdtimeout=300; Terminate call if 300 seconds of no RTP activity useragent=dhaval ; Allows you to change the user agent string dtmfmode = rfc2833; Set default dtmfmode for sending DTMF. Default: rfc2833 qualify=yes nat=yes ;canreinvite=yes directmedia=yes directrtpsetup=yes And here is DB fields snapshots. id: 1 name: 201 ipaddr: 172.18.100.243 port: 53624 regseconds: 1328716180 defaultuser: 201 fullcontact: NULL regserver: dhaval useragent: CSipSimple r1133 / b lastms: 554 host: dynamic type: friend context: from-internal permit: NULL deny: NULL secret: 201 md5secret: NULL remotesecret: NULL transport: NULL dtmfmode: NULL directmedia: yes nat: NULL allow: ulaw disallow: g729 insecure: invite callerid: NULL rfc2833compensate: NULL mailbox: NULL session-timers: NULL session-expires: NULL session-minse: NULL session-refresher: NULL Kindly help me to resolve this. Thanks Dhaval The first thing I would try is 'rtcachefriends=yes', that should do it. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by
[asterisk-users] error during dahdi installation on centos
Hello, # rpm -qa | grep kernel kernel-headers-2.6.18-274.18.1.el5 kernel-PAE-2.6.18-128.el5 kernel-devel-2.6.18-274.18.1.el5 kernel-PAE-devel-2.6.18-274.18.1.el5 [root@localhost ~]# uname -i i386 Trying to install dahdi-linux-complete-2.3.0.1+2.3.0 on CentOS but get below error. Can you please assist in this? [root@localhost dahdi-linux-complete-2.3.0.1+2.3.0]# make make -C linux all make[1]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-128.el5PAE kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 Thanks, Kamlesh -- _ -- 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] error during dahdi installation on centos
On Wed, Feb 15, 2012 at 12:11:18PM +, Kamlesh Kumar wrote: Hello, # rpm -qa | grep kernel kernel-headers-2.6.18-274.18.1.el5 kernel-PAE-2.6.18-128.el5 kernel-devel-2.6.18-274.18.1.el5 kernel-PAE-devel-2.6.18-274.18.1.el5 [root@localhost ~]# uname -i i386 Trying to install dahdi-linux-complete-2.3.0.1+2.3.0 on CentOS but get below error. Can you please assist in this? [root@localhost dahdi-linux-complete-2.3.0.1+2.3.0]# make make -C linux all make[1]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-128.el5PAE kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 Boot to the newer kernel and/or use: make KVERS=2.6.18-274.18.1.el5PAE -- 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] error during dahdi installation on centos
thank you very much for your quick response. make KVERS=2.6.18-274.18.1.el5PAE It started the installation but stuck at below error LD [M] /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/wcte12xp/wcte12xp.o CC [M] /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp/card_bri.o In file included from /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp/xpd.h:31, from /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp/card_bri.c:29: include/linux/device.h:407: error: expected identifier or â(â before âconstâ make[4]: *** [/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp/card_bri.o] Error 1 make[3]: *** [/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/xpp] Error 2 make[2]: *** [_module_/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.18-274.18.1.el5-PAE-i686' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 Regards, Kamlesh Date: Wed, 15 Feb 2012 14:39:00 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] error during dahdi installation on centos On Wed, Feb 15, 2012 at 12:11:18PM +, Kamlesh Kumar wrote: Hello, # rpm -qa | grep kernel kernel-headers-2.6.18-274.18.1.el5 kernel-PAE-2.6.18-128.el5 kernel-devel-2.6.18-274.18.1.el5 kernel-PAE-devel-2.6.18-274.18.1.el5 [root@localhost ~]# uname -i i386 Trying to install dahdi-linux-complete-2.3.0.1+2.3.0 on CentOS but get below error. Can you please assist in this? [root@localhost dahdi-linux-complete-2.3.0.1+2.3.0]# make make -C linux all make[1]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-128.el5PAE kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 Boot to the newer kernel and/or use: make KVERS=2.6.18-274.18.1.el5PAE -- 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 -- _ -- 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] Forwarding queue to remote agent over PSTN
Hi, A client is looking for a way to have queue agents available over their mobile or land-line phones. In other words, some queue members would be local (over SIP channels) while others would only be reachable by dialling their (mobile) phones over the PSTN. Is there some easy way to accomplish this? Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] OT - Which cheap ISDN phone
Hello, For backup, I'm looking after cheap ISDN phones I could use in BRI PtP or PtmP lines to forward incoming calls (typing a DTMF sequence). Which model and brand would you recommend for this ? Regards -- _ -- 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] Forwarding queue to remote agent over PSTN
You could register the agent to a SIP extension with followme. When the queue went to ring the SIP extension, followme would send the call on to the mobile/land line. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Raj Mathur (??? ?) Sent: Wednesday, February 15, 2012 7:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Forwarding queue to remote agent over PSTN Hi, A client is looking for a way to have queue agents available over their mobile or land-line phones. In other words, some queue members would be local (over SIP channels) while others would only be reachable by dialling their (mobile) phones over the PSTN. Is there some easy way to accomplish this? Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] Forwarding queue to remote agent over PSTN
On 15-02-12 14:31, Danny Nicholas wrote: You could register the agent to a SIP extension with followme. When the queue went to ring the SIP extension, followme would send the call on to the mobile/land line. It's been decades since I last bought an ISDN phone and I'm not even sure you can still buy those. At least I have not seen those in .nl in a long time. Maybe anything from eBay that works? Regards, Patrick -- _ -- 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 + Avaya (CM5.2) H.323 trunk Link (Dustin fails)
Can be done calls from each system? How about to capture data with Wireshark? I have experiences Asterisk with Panasonic with H323 without any problem. Mc GRATH Ricardo E-Mail mcgra...@mail2web.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] Polycom firmware 4.0.1 and paging
Which version of asterisk are you using? I just have this in 1.4 and it works fine: SIPAddHeader(Alert-Info: intercom); -Dave On 02/14/2012 08:10 PM, Mike wrote: In case anybody was following this thread, or someone Googles it in the future, here is the solution: This worked fine with Polycom firmware 3.3x: exten = s,n,SIPAddHeader(Alert-Info:Ring Answer) For firmware 4.0+, apparently I needed to add info=, i.e.: exten = s,n,SIPAddHeader(Alert-Info: info=Ring Answer) Simple, yet quite obscure (for me at least). Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, February 13, 2012 10:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging Thanks Dave, it at least gives me hope that my efforts aren`t wasted. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Monday, February 13, 2012 9:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging On 02/10/2012 05:30 PM, Mike wrote: Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer? I would second the others suggestions about rewriting the configs. Polycom made extensive changes between 3.2 and 3.3, and I think they made a fair number of changes between 3.3 and 4.0. I have two phones that I've upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I believe I have auto answer working as you describe. Here's the pertinent snippet from my config: polycomConfig voIpProt voIpProt.SIP voIpProt.SIP.alertInfo voIpProt.SIP.alertInfo.1.class=ringAutoAnswer voIpProt.SIP.alertInfo.1.value=intercom voIpProt.SIP.alertInfo.2.class=ringAnswerMute voIpProt.SIP.alertInfo.2.value=page voIpProt.SIP.alertInfo.3.class=autoAnswer voIpProt.SIP.alertInfo.3.value=silentanswer /voIpProt.SIP.alertInfo /voIpProt.SIP /voIpProt /polycomConfig I have also added anse.rt section to adjust the ringer and timeouts for these ring tones. -Dave -- _ -- 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] OT - T.38 unreliable on a LAN : truth or obscurantism ?
Hi, When someone says T.38 is not reliable on a (normally loaded and managed) LAN, would you rather agree or disagree ? In this case, fax calls are coming in through an analog gateway, passing trough Asterisk and then going out to ISDN through a digital gateway. Comments ? Regards -- _ -- 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] OT - T.38 unreliable on a LAN : truth or obscurantism ?
- Original Message - Hi, When someone says T.38 is not reliable on a (normally loaded and managed) LAN, would you rather agree or disagree ? In this case, fax calls are coming in through an analog gateway, passing trough Asterisk and then going out to ISDN through a digital gateway. Is T.38 actually in use in this scenario? Or are you simply passing the fax call through Asterisk as 'normal' audio (G.711u/a, etc)? If so, you may want to see here: http://www.soft-switch.org/foip.html --Tim -- _ -- 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] OT - T.38 unreliable on a LAN : truth or obscurantism ?
T.38 is tolerant of most network conditions, ... the challenges in getting reliable performance are usually limited to getting the interop right once, but the absolute success rate will depend on the quality of your T.38/PSTN gateway's fax implementation. In general terms, T.38 is actually the right way to cope with lossy or high jitter network conditions, and so it's reliable over most networks. The question people usually ask is whether fax over G.711 is unreliable on a LAN. To which the answer would be a definite 'it depends' ;-) -d On Feb 15, 2012, at 3:03 PM, Olivier wrote: Hi, When someone says T.38 is not reliable on a (normally loaded and managed) LAN, would you rather agree or disagree ? In this case, fax calls are coming in through an analog gateway, passing trough Asterisk and then going out to ISDN through a digital gateway. Comments ? Regards -- _ -- 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] OT - T.38 unreliable on a LAN : truth or obscurantism ?
2012/2/15, Tim Nelson tnel...@rockbochs.com: - Original Message - Hi, When someone says T.38 is not reliable on a (normally loaded and managed) LAN, would you rather agree or disagree ? In this case, fax calls are coming in through an analog gateway, passing trough Asterisk and then going out to ISDN through a digital gateway. Is T.38 actually in use in this scenario? Or are you simply passing the fax call through Asterisk as 'normal' audio (G.711u/a, etc)? Yes, T.38 is in use between each gateway and Asterisk (I should have specified this more clearly) : Fax Machine -- Analog Gw --T.38-- Asterisk --T.38 -- Digital Gw --ISDN-- PSTN If so, you may want to see here: http://www.soft-switch.org/foip.html --Tim -- _ -- 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] OT - T.38 unreliable on a LAN : truth or obscurantism ?
2012/2/15, Darren Nickerson darren.nicker...@ifax.com: T.38 is tolerant of most network conditions, ... the challenges in getting reliable performance are usually limited to getting the interop right once, but the absolute success rate will depend on the quality of your T.38/PSTN gateway's fax implementation. In general terms, T.38 is actually the right way to cope with lossy or high jitter network conditions, and so it's reliable over most networks. Yes. An other thing to factor in, is how Asterisk's load could influence its capability to let faxes passing through. To me, if Asterisk is installed on a modern CPU (dual core and more) and is configured in such a way that no transcoding happen, then passing faxes through is easy and works reliably. Opinions ? The question people usually ask is whether fax over G.711 is unreliable on a LAN. To which the answer would be a definite 'it depends' ;-) -d On Feb 15, 2012, at 3:03 PM, Olivier wrote: Hi, When someone says T.38 is not reliable on a (normally loaded and managed) LAN, would you rather agree or disagree ? In this case, fax calls are coming in through an analog gateway, passing trough Asterisk and then going out to ISDN through a digital gateway. Comments ? Regards -- _ -- 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 -- _ -- 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] OT - T.38 unreliable on a LAN : truth or obscurantism ?
On Feb 15, 2012, at 4:03 PM, Olivier wrote: 2012/2/15, Darren Nickerson darren.nicker...@ifax.com: T.38 is tolerant of most network conditions, ... the challenges in getting reliable performance are usually limited to getting the interop right once, but the absolute success rate will depend on the quality of your T.38/PSTN gateway's fax implementation. In general terms, T.38 is actually the right way to cope with lossy or high jitter network conditions, and so it's reliable over most networks. Yes. An other thing to factor in, is how Asterisk's load could influence its capability to let faxes passing through. To me, if Asterisk is installed on a modern CPU (dual core and more) and is configured in such a way that no transcoding happen, then passing faxes through is easy and works reliably. Opinions ? The devil is in the details, but in general it's nowhere near that simple. You don't clarify what pass-through role Asterisk is playing here. G.711? T.38? What are you passing through TO? A TDM card connected to the PSTN? Or some SIP trunking provider, who themselves may be using G.711 or T.38 ... Assuming you mean the specific case of one local LAN hop over SIP, connecting directly to a well-configured PSTN card on the same Asterisk server, it's possible to get reliable faxing over G.711 with careful network configuration, good and well configured ethernet interfaces, correct jitter buffer, gain and echo cancelation settings, etc etc. -d -- _ -- 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] OT - T.38 unreliable on a LAN : truth or obscurantism ?
On Feb 15, 2012, at 3:49 PM, Olivier wrote: 2012/2/15, Tim Nelson tnel...@rockbochs.com: - Original Message - Hi, When someone says T.38 is not reliable on a (normally loaded and managed) LAN, would you rather agree or disagree ? In this case, fax calls are coming in through an analog gateway, passing trough Asterisk and then going out to ISDN through a digital gateway. Is T.38 actually in use in this scenario? Or are you simply passing the fax call through Asterisk as 'normal' audio (G.711u/a, etc)? Yes, T.38 is in use between each gateway and Asterisk (I should have specified this more clearly) : Fax Machine -- Analog Gw --T.38-- Asterisk --T.38 -- Digital Gw --ISDN-- PSTN Assuming you have Asterisk doing T.38 pass-through here, reinviting the T.38 payload to go directly between the analog GW and the Digital GW, and assuming that 'Digital Gw' has a good T.30 fax engine inside of it (because after all, the gateway is what's speaking convention audio-based fax to the remote sender/receiver, the above setup should work well independent of network conditions. T.38 has ways of coping with extremely bad connections (via packet redundancy or FEC error correction) that you probably would not need on a LAN. Note, however, the use of T.38 versus G.711 may limit the speed of your faxing to 14,400 and prevent the fax protocol from using its own error correction (many T.38 gateway implementations wrong-headedly disable ECM error correction). When it comes to faxing over a LAN, the choice of T.38 versus G.711 uLaw/aLaw is less than obvious. In your case, it will be highly dependent upon each piece of your call flow. The fax machine, the analog gateway, how Asterisk is setup, the digital gateway and the quality of the PSTN line. These days you cannot trust that your PSTN carrier is using TDM routes, sometimes they slip a little T.38 in the middle on you, and all bets are off. No matter what scenario you go with though, you probably want to get Asterisk out of the media path and get a gateway-to-gateway conversation going eventually. -Darren -- _ -- 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] chan_capi audio weirdness
Hello Arik, On 02/14/2012 12:49 PM, Arik Raffael Funke wrote: Hi, I am trying to run asterisk on an AVM Fritz!Box Fon 7270 embedded DSL router. This works quite well after getting rid of the preinstalled phone server but I am encountering some unexpected behaviour. Background: I am using two CAPI controllers provided by the hardware - one in MSN mode for dialling out and - one in NT-mode, (DID) for the internal S0-Bus The problem is, I get no audio whatsoever until a channel is answered. Some of the symptoms of this are: - If I have an s-extension for the internal S0-Bus exten = s,1,Playtones(dial) I cannot hear the dialtone. It works however with: exten = s,1,Answer exten = s,n,Playtones(dial) - Similarly if I dial from internal to external with the extension: exten = _X.,1,Dial(CAPI/contr1/12345) I hear no progress indication. EVEN when using the r-option of the dial command. It works however with exten = _X.,1,Answer exten = _X.,n,Dial(CAPI/contr1/12345) in NT mode, the B-channel is not activated automatically. You have to signal the TE side that early-B3 data is available. Then the TE side can activate the B-channel. If the NT-side is chan_capi, use exten = _X.,n,capicommand(progress) without Answer before Dial(). Also, when using Dial() with chan_capi, you should use /b or /B option in Dial() to get early-B3 from that other side too. See README of chan_capi package for more details. Armin -- _ -- 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] Call holding with chan_capi
Hi,
On 02/14/2012 06:28 PM, Arik Raffael Funke wrote:
My apologies, I just realised I copied the wrong section of the debug log. So
once again, when pressing the park call button, I get the following capi
debug output:
CAPI: ApplId=0x0002 Command=0x80 SubCommand=0x82 MsgNum=0xe446 NCCI=0x1403
FACILITY_IND ID=002 #0xe446 LEN=0018
Controller/PLCI/NCCI= 0x1403
FacilitySelector= 0x3
FacilityIndicationParameter = 02 80 00
-- ISDN_INTERN#02: unhandled FACILITY_IND supplementary function 8002
FACILITY_RESP ID=002 #0xe446 LEN=0015
Controller/PLCI/NCCI= 0x1403
FacilitySelector= 0x3
FacilityResponseParameters = default
CAPI: ApplId=0x0002 Command=0x84 SubCommand=0x82 MsgNum=0xe447 NCCI=0x00011403
DISCONNECT_B3_IND ID=002 #0xe447 LEN=0015
Controller/PLCI/NCCI= 0x11403
Reason_B3 = 0x3301
NCPI= default
DISCONNECT_B3_RESP ID=002 #0xe447 LEN=0012
Controller/PLCI/NCCI= 0x11403
looks like normal hold where B-channel is released.
When you use capicommand(hold), you can specify a second parameter. This
parameter is the name of a variable which is filled with the reference ID of
the hold. capicommand(retrieve, ${HOLDID}) then can unhold the call.
See README of chan_capi package for details.
Armin
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Re: [asterisk-users] Polycom firmware 4.0.1 and paging
With Polycom firmware 4.0.1b? I have 1.8, one of the latest can`t remember which is installed on that server. Maybe the fact that my alert info has two words, and isn`t parsed correctly by Polycom...? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Wednesday, February 15, 2012 10:20 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging Which version of asterisk are you using? I just have this in 1.4 and it works fine: SIPAddHeader(Alert-Info: intercom); -Dave On 02/14/2012 08:10 PM, Mike wrote: In case anybody was following this thread, or someone Googles it in the future, here is the solution: This worked fine with Polycom firmware 3.3x: exten = s,n,SIPAddHeader(Alert-Info:Ring Answer) For firmware 4.0+, apparently I needed to add info=, i.e.: exten = s,n,SIPAddHeader(Alert-Info: info=Ring Answer) Simple, yet quite obscure (for me at least). Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, February 13, 2012 10:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging Thanks Dave, it at least gives me hope that my efforts aren`t wasted. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Monday, February 13, 2012 9:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging On 02/10/2012 05:30 PM, Mike wrote: Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer? I would second the others suggestions about rewriting the configs. Polycom made extensive changes between 3.2 and 3.3, and I think they made a fair number of changes between 3.3 and 4.0. I have two phones that I've upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I believe I have auto answer working as you describe. Here's the pertinent snippet from my config: polycomConfig voIpProt voIpProt.SIP voIpProt.SIP.alertInfo voIpProt.SIP.alertInfo.1.class=ringAutoAnswer voIpProt.SIP.alertInfo.1.value=intercom voIpProt.SIP.alertInfo.2.class=ringAnswerMute voIpProt.SIP.alertInfo.2.value=page voIpProt.SIP.alertInfo.3.class=autoAnswer voIpProt.SIP.alertInfo.3.value=silentanswer /voIpProt.SIP.alertInfo /voIpProt.SIP /voIpProt /polycomConfig I have also added anse.rt section to adjust the ringer and timeouts for these ring tones. -Dave -- _ -- 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] OT - T.38 unreliable on a LAN : truth or obscurantism ?
On 02/15/2012 03:03 PM, Olivier wrote: Hi, When someone says T.38 is not reliable on a (normally loaded and managed) LAN, would you rather agree or disagree ? In this case, fax calls are coming in through an analog gateway, passing trough Asterisk and then going out to ISDN through a digital gateway. Comments ? While I can't speak for Asterisk's T.38 performance (it was barely past the point of okay, it compiles at the time of this datapoint), T.38 in general can handle nasty network conditions without a problem as long you enable some sort of error correction (either FEC or packet redundancy). Case in point, I ran several hundred SIP-based T.38 calls a month over VSAT links. The links ran anywhere from 550 to 750ms latency and would average around 1-2% packet loss (averaged over a 5 minute period). Those were with a Quintum ASG400 at the far end and a Quintum CMS960 going into PRIs at the VSAT hub. So if T.38 can handle that, it can handle just about anything. -- _ -- 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] Block Collect Calls on ISDN trunk
How to block collect calls on ISDN trunk? Thank's Att, Rafael Saraiva -- _ -- 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] Polycom firmware 4.0.1 and paging
It appears you need the info= if the string you are using is enclosed in angle brackets. Alert-Info: fooworks Alert-Info:foo does not work Alert-Info:info=foo works On Wed, Feb 15, 2012 at 2:09 PM, Mike l...@net-wall.com wrote: With Polycom firmware 4.0.1b? I have 1.8, one of the latest can`t remember which is installed on that server. Maybe the fact that my alert info has two words, and isn`t parsed correctly by Polycom...? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Wednesday, February 15, 2012 10:20 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging Which version of asterisk are you using? I just have this in 1.4 and it works fine: SIPAddHeader(Alert-Info: intercom); -Dave On 02/14/2012 08:10 PM, Mike wrote: In case anybody was following this thread, or someone Googles it in the future, here is the solution: This worked fine with Polycom firmware 3.3x: exten = s,n,SIPAddHeader(Alert-Info:Ring Answer) For firmware 4.0+, apparently I needed to add info=, i.e.: exten = s,n,SIPAddHeader(Alert-Info: info=Ring Answer) Simple, yet quite obscure (for me at least). Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, February 13, 2012 10:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging Thanks Dave, it at least gives me hope that my efforts aren`t wasted. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Monday, February 13, 2012 9:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging On 02/10/2012 05:30 PM, Mike wrote: Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer? I would second the others suggestions about rewriting the configs. Polycom made extensive changes between 3.2 and 3.3, and I think they made a fair number of changes between 3.3 and 4.0. I have two phones that I've upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I believe I have auto answer working as you describe. Here's the pertinent snippet from my config: polycomConfig voIpProt voIpProt.SIP voIpProt.SIP.alertInfo voIpProt.SIP.alertInfo.1.class=ringAutoAnswer voIpProt.SIP.alertInfo.1.value=intercom voIpProt.SIP.alertInfo.2.class=ringAnswerMute voIpProt.SIP.alertInfo.2.value=page voIpProt.SIP.alertInfo.3.class=autoAnswer voIpProt.SIP.alertInfo.3.value=silentanswer /voIpProt.SIP.alertInfo /voIpProt.SIP /voIpProt /polycomConfig I have also added anse.rt section to adjust the ringer and timeouts for these ring tones. -Dave -- _ -- 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 -- _ -- 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] Block Collect Calls on ISDN trunk
How to block collect calls on ISDN trunk? You need Asterisk v1.8 or later and check the value of CHANNEL(reversecharge) in your dialplan. https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL Richard -- _ -- 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] Block Collect Calls on ISDN trunk
Richard Can you give me an example of how to use this function? Att, Rafael Saraiva 2012/2/15 Richard Mudgett rmudg...@digium.com How to block collect calls on ISDN trunk? You need Asterisk v1.8 or later and check the value of CHANNEL(reversecharge) in your dialplan. https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL Richard -- _ -- 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] Block Collect Calls on ISDN trunk
How to block collect calls on ISDN trunk?
You need Asterisk v1.8 or later and check the value of
CHANNEL(reversecharge) in your dialplan.
https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL
Can you give me an example of how to use this function?
exten = 100,1,Proceeding()
same = n,GotoIf($[${CHANNEL(reversecharge)} = -1]?allow:block)
same = n(allow),Dial()
same = n(block),Hangup()
Please note that CHANNEL(reversecharge) is only valid on ISDN channels.
Richard
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[asterisk-users] Dahdi Answer a Call On ringing State.
Hi, I have setup Dahdi with Sangoma FXO A200 card and asterisk 1.8 , everything seems fine and working perfectly incoing/outgoing. but one major issue is, when i made an out call from dahdi trunks and when a number is in ringing state it gives me an answer state. so i cannot develop any custom application which can use a screening macro because when a cellphone is in ringing state call is answered by dahdi channel so it will start executing dial plan which causes an issue. let me know if there is any parameter from which i can set in chan_dahdi.conf and check if it worked or not. Note: I am from INDIA and line is from BSNL thanks Dhaval -- _ -- 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] Forwarding queue to remote agent over PSTN
If you have your agents static(hard coded) for Queue in queues.conf, you could add following in your queue definition, member = DAHDI/G0/XX Replace XX with your Agent's cellphone or Landline number. And if you have your Agents added dynamically in Queue, use local channel as a Queue member and have your local channel dial the cellphone or Landline number. See the 'Using Local Channels' section on a link http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html for more information. (Courtesy:Leif Madsen, Jim Van Meggelen, and Russell Bryant) --Satish Barot On Wed, Feb 15, 2012 at 6:32 PM, Raj Mathur (राज माथुर) r...@linux-delhi.org wrote: Hi, A client is looking for a way to have queue agents available over their mobile or land-line phones. In other words, some queue members would be local (over SIP channels) while others would only be reachable by dialling their (mobile) phones over the PSTN. Is there some easy way to accomplish this? Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] Forwarding queue to remote agent over PSTN
On Thursday 16 Feb 2012, Satish Barot wrote: If you have your agents static(hard coded) for Queue in queues.conf, you could add following in your queue definition, member = DAHDI/G0/XX Replace XX with your Agent's cellphone or Landline number. And if you have your Agents added dynamically in Queue, use local channel as a Queue member and have your local channel dial the cellphone or Landline number. See the 'Using Local Channels' section on a link http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html for more information. (Courtesy:Leif Madsen, Jim Van Meggelen, and Russell Bryant) That's brilliant, and answers all my questions. Thanks! Also thanks to Danny Nicholas for the initial push in the right direction. Regards, -- Raj On Wed, Feb 15, 2012 at 6:32 PM, Raj Mathur (राज माथुर) r...@linux-delhi.org wrote: A client is looking for a way to have queue agents available over their mobile or land-line phones. In other words, some queue members would be local (over SIP channels) while others would only be reachable by dialling their (mobile) phones over the PSTN. Is there some easy way to accomplish this? -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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
