Re: [asterisk-users] Building Asterisk 11.4.0-rc1 with PJSIP 2.1
hi, i would try to make a symlink... link the wrong folder to the correct one... yves Am 02.05.2013 23:34, schrieb James Mortensen: Hello, I'm working on building Asterisk 11.4.0-rc1 with pjproject 2.1 instead of 2.0 due to a crashing issue resulting from ICE. https://issues.asterisk.org/jira/browse/ASTERISK-21696 Currently, I'm systematically going through each Makefile in every directory in pjproject and changing the paths that exist in the pjproject 2.0 included with Asterisk, so that I can successfully build Asterisk. I'm using the Asterisk pjproject 2.1 port from here: https://github.com/asterisk/pjproject An example of the build errors I'm resolving one by one is this: make[2]: *** No rule to make target `../../pjlib/lib/libpj-x86_64-unknown-linux-gnu.a', needed by `../lib/libpjnath-x86_64-unknown-linux-gnu.a'. Stop. make[1]: *** [/mnt/src/asterisk-11.4.0-rc1/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a] Error 2 make: *** [res] Error 2 I'm editing the Makefiles and fixing the paths so Asterisk can find the target. For all the people out there smarter than me, is there a better way to go about this? I'm hoping upgrading PJSIP will resolve the crashing issue, and I'll continue going through Makefiles until someone smarter than me can enlighten me. Thank you for your help! -- James Mortensen Project Manager, VoiceCurve, Inc. 866-707-4590 james.morten...@voicecurve.com mailto:james.morten...@voicecurve.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 -- _ -- 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] VoIP Incoming Issue
I have made the SIP bind port to 5070, and already I have one VoIP trunk configured in my Asterisk 1.6. Now the problem is after changing the bind port at some point of time, am not able to dial in the DID number of the VoIP trunk! Changing the bind port matters 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] debug strategy for one-way audio calls
@Marrie For one way audio as a debug strategy you can enable RTP debug and see whether you have both way packets flow SENT and GOT. Regards On Thu, May 2, 2013 at 6:05 PM, Johan Wilfer li...@jttech.se wrote: 2013-05-02 13:19, Marie Fischer skrev: Hello everybody, from time to time, we get so-called simplex / one-way audio calls, where one party cannot hear the other. The only thing in common is that is does happen with calls via SIP trunk, not ISDN and not internal calls. Nothing strange in verbose and SIP logs. Could even be some weird intermittent firewall issue I guess. Apart from logging all traffic 24/7 via tcpdump (not really convenient), can you give me some ideas how to debug this kind of issue? Asterisk 1.8.11-cert10 on CentOS 6.3, if that matters. Voipmonitor.org is great for debugging voip. You can either use only the sniffer (opensource) and use mysql + the pcap files or you can also buy the commercial webgui. Either way, it's a great product. /Johan -- _ -- 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 QSIG doesnt send the calling name to Nortel CS1000
Thanks Dale for your answer. I am a consultant for a project for Banca D'Italia and we have to give birth to 20 branches with 1,200 snom phones ... oh my god I can not record 1200 names on Nortel!!! : '( Il giorno giovedì 2 maggio 2013, Dale Noll dn...@wi.rr.com ha scritto: I retired my Nortel switch a couple of years ago, but I do not believe I ever got Asterisk - Nortel to pass the CPND, just the number. If I remember correctly, I had to enter then names manually in Nortel (LD 95?) for display on the Nortel endpoints. On Tue, Apr 30, 2013 at 11:30 AM, Danilo Dionisi dionisi.dan...@gmail.com wrote: Hello to all, I have a problem with an asterisk qsig. I have three machines: Nortel CS1000 --- Card Sangoma PRI --- Asterisk QSIG ---SIP Trunk--- Asterisk I use Snom phones on Asterisk. If I call from Asterisk to Nortel, Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone, but if I call from Nortel to Asterisk, the QSIG does not send Nortel on the display of the name of the person i'm calling ... why? example: Snom phone = Danilo 1001 Nortel phone = Marco 2002 If I call from Nortel to Asterisk, I have the display of the Snom Marco 2002 and the display of Nortel Danilo 1001; If I call from Nortel to Asterisk, I have the display of the Snom Marco 2002 and the display of Nortel 1001 This is my / etc / asterisk / chan_dahdi.conf [channels] cc_offer_timer=20 ccbs_available_timer=4800 ccnr_available_timer=7200 cc_recall_timer=20 cc_agent_policy=native cc_monitor_policy=native pridialplan=private prilocaldialplan=private context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no facilityenable=yes callerid=asreceived ;Sangoma A104 port 1 [slot:4 bus:17 span:1] wanpipe1 switchtype=qsig context=from_nortel group=0 echocancel=yes faxdetect=incoming signalling=pri_cpe channel =1-15,17-31 -- _ -- 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 QSIG doesnt send the calling name to Nortel CS1000
I'm sorry, the mail is automatically send :p However, I am for the Asterisk, there are other external consultants for Nortel ... according to you can be out a patch for Asterisk to send the facility of CPND??? Danilo Il giorno venerdì 3 maggio 2013, Danilo Dionisi dionisi.dan...@gmail.com ha scritto: Thanks Dale for your answer. I am a consultant for a project for Banca D'Italia and we have to give birth to 20 branches with 1,200 snom phones ... oh my god I can not record 1200 names on Nortel!!! : '( Il giorno giovedì 2 maggio 2013, Dale Noll dn...@wi.rr.com ha scritto: I retired my Nortel switch a couple of years ago, but I do not believe I ever got Asterisk - Nortel to pass the CPND, just the number. If I remember correctly, I had to enter then names manually in Nortel (LD 95?) for display on the Nortel endpoints. On Tue, Apr 30, 2013 at 11:30 AM, Danilo Dionisi dionisi.dan...@gmail.com wrote: Hello to all, I have a problem with an asterisk qsig. I have three machines: Nortel CS1000 --- Card Sangoma PRI --- Asterisk QSIG ---SIP Trunk--- Asterisk I use Snom phones on Asterisk. If I call from Asterisk to Nortel, Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone, but if I call from Nortel to Asterisk, the QSIG does not send Nortel on the display of the name of the person i'm calling ... why? example: Snom phone = Danilo 1001 Nortel phone = Marco 2002 If I call from Nortel to Asterisk, I have the display of the Snom Marco 2002 and the display of Nortel Danilo 1001; If I call from Nortel to Asterisk, I have the display of the Snom Marco 2002 and the display of Nortel 1001 This is my / etc / asterisk / chan_dahdi.conf [channels] cc_offer_timer=20 ccbs_available_timer=4800 ccnr_available_timer=7200 cc_recall_timer=20 cc_agent_policy=native cc_monitor_policy=native pridialplan=private prilocaldialplan=private context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no facilityenable=yes callerid=asreceived ;Sangoma A104 port 1 [slot:4 bus:17 span:1] wanpipe1 switchtype=qsig context=from_nortel group=0 echocancel=yes faxdetect=incoming signalling=pri_cpe channel =1-15,17-31 -- _ -- 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 QSIG doesnt send the calling name to Nortel CS1000
I never researched the problem deep enough to determine if the problem was Asterisk's QSIG implementation or Nortel's. I had several Nortel switches that could pass CPND to each other, but none of them were using QSIG. I do not remember what switchtype they were set up as... DM100 maybe? I did not do this for my project, but in theory, a script could be written to connect the Nortel system via Ethernet (rlogin protocol), login, start the load and update the names database. I did script some other things for that. I will have to look for my documentation on the connection. On Fri, May 3, 2013 at 4:27 AM, Danilo Dionisi dionisi.dan...@gmail.comwrote: I'm sorry, the mail is automatically send :p However, I am for the Asterisk, there are other external consultants for Nortel ... according to you can be out a patch for Asterisk to send the facility of CPND??? Danilo Il giorno venerdì 3 maggio 2013, Danilo Dionisi dionisi.dan...@gmail.com ha scritto: Thanks Dale for your answer. I am a consultant for a project for Banca D'Italia and we have to give birth to 20 branches with 1,200 snom phones ... oh my god I can not record 1200 names on Nortel!!! : '( Il giorno giovedì 2 maggio 2013, Dale Noll dn...@wi.rr.com ha scritto: I retired my Nortel switch a couple of years ago, but I do not believe I ever got Asterisk - Nortel to pass the CPND, just the number. If I remember correctly, I had to enter then names manually in Nortel (LD 95?) for display on the Nortel endpoints. On Tue, Apr 30, 2013 at 11:30 AM, Danilo Dionisi dionisi.dan...@gmail.com wrote: Hello to all, I have a problem with an asterisk qsig. I have three machines: Nortel CS1000 --- Card Sangoma PRI --- Asterisk QSIG ---SIP Trunk--- Asterisk I use Snom phones on Asterisk. If I call from Asterisk to Nortel, Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone, but if I call from Nortel to Asterisk, the QSIG does not send Nortel on the display of the name of the person i'm calling ... why? example: Snom phone = Danilo 1001 Nortel phone = Marco 2002 If I call from Nortel to Asterisk, I have the display of the Snom Marco 2002 and the display of Nortel Danilo 1001; If I call from Nortel to Asterisk, I have the display of the Snom Marco 2002 and the display of Nortel 1001 This is my / etc / asterisk / chan_dahdi.conf [channels] cc_offer_timer=20 ccbs_available_timer=4800 ccnr_available_timer=7200 cc_recall_timer=20 cc_agent_policy=native cc_monitor_policy=native pridialplan=private prilocaldialplan=private context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no facilityenable=yes callerid=asreceived ;Sangoma A104 port 1 [slot:4 bus:17 span:1] wanpipe1 switchtype=qsig context=from_nortel group=0 echocancel=yes faxdetect=incoming signalling=pri_cpe channel =1-15,17-31 -- _ -- 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] Asterisk QSIG doesnt send the calling name to Nortel CS1000
Hello to all, I have a problem with an asterisk qsig . I have three machines : Nortel CS1000 --- Card Sangoma PRI --- Asterisk QSIG --- SIP Trunk --- Asterisk I use Snom phones on Asterisk . If I call from Asterisk to Nortel , Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone , but if I call from Nortel to Asterisk , the QSIG does not send Nortel on the display of the name of the person i'm calling ... why? example: Snom phone = Danilo 1001 Nortel phone = Marco 2002 If I call from Nortel to Asterisk , I have the display of the Snom Marco 2002 and the display of Nortel Danilo 1001 ; If I call from Nortel to Asterisk , I have the display of the Snom Marco 2002 and the display of Nortel 1001 Try placing a Wait(1) in the dialplan for the Nortel to Asterisk direction. Many Q.SIG implementations send the name in a separate message AFTER sending the SETUP message. Asterisk usually puts the call into dialplan when it receives the SETUP message. Waiting allows a subsequent message containing the name to arrive and be available in the dialplan for subsequent outward dials. 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] Asterisk QSIG doesnt send the calling name to Nortel CS1000
Really, you can update the Nortel database of names with a script? *.* In what language it is possible to write the script? Danilo Il giorno venerdì 3 maggio 2013, Richard Mudgett rmudg...@digium.com ha scritto: Hello to all, I have a problem with an asterisk qsig . I have three machines : Nortel CS1000 --- Card Sangoma PRI --- Asterisk QSIG --- SIP Trunk --- Asterisk I use Snom phones on Asterisk . If I call from Asterisk to Nortel , Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone , but if I call from Nortel to Asterisk , the QSIG does not send Nortel on the display of the name of the person i'm calling ... why? example: Snom phone = Danilo 1001 Nortel phone = Marco 2002 If I call from Nortel to Asterisk , I have the display of the Snom Marco 2002 and the display of Nortel Danilo 1001 ; If I call from Nortel to Asterisk , I have the display of the Snom Marco 2002 and the display of Nortel 1001 Try placing a Wait(1) in the dialplan for the Nortel to Asterisk direction. Many Q.SIG implementations send the name in a separate message AFTER sending the SETUP message. Asterisk usually puts the call into dialplan when it receives the SETUP message. Waiting allows a subsequent message containing the name to arrive and be available in the dialplan for subsequent outward dials. 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] Asterisk QSIG doesnt send the calling name to Nortel CS1000
It is not something that Nortel ever really talked about. I had used their tool Meridian Admin Tool (MAT) which used an Ethernet connection to my switches to sent commands over the network. It did not take much to figure out that they we using an rlogin connection(tcpdump and wire shark are your friends). From that point forward, I could simply use rlogin to issue commands to the switches right from my Linux command prompt. The next step was to automate commands using the available tools. I have used Perl and Expect for that. Perl being my favorite, but Expect has some nice features for this. Your choice. There is a username required for the rlogion connection. I do not know if that username is unique per customer or generic to all systems. The command line I used to get a Nortel command prompt was: rlogin nortelswitch -l CPSID1110 -e'%' This site says the name is simply CPSID. http://blog.michaelfmcnamara.com/2008/04/how-to-rlogin-to-a-nortel-call-server/ Now, back to the original topic. Perhaps I did not read your original post correctly. Is your problem that... A: Calls from Asterisk to Nortel do not display Asterisk names on the Nortel phones or B: Calls from Nortel to Asterisk do not display Nortel names on the SMON phones My problem was A and matching the outbound CLID from Asterisk with a name entry on Nortel fixed the problem. If your problem is B, I would go with Richard's suggestion. I do not think that I had problem B, but I do database dips for CLID lookup on inbound calls so I may not have noticed the problem. Dale On Fri, May 3, 2013 at 10:55 AM, Danilo Dionisi dionisi.dan...@gmail.comwrote: Really, you can update the Nortel database of names with a script? *.* In what language it is possible to write the script? Danilo Il giorno venerdì 3 maggio 2013, Richard Mudgett rmudg...@digium.com ha scritto: Hello to all, I have a problem with an asterisk qsig . I have three machines : Nortel CS1000 --- Card Sangoma PRI --- Asterisk QSIG --- SIP Trunk --- Asterisk I use Snom phones on Asterisk . If I call from Asterisk to Nortel , Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone , but if I call from Nortel to Asterisk , the QSIG does not send Nortel on the display of the name of the person i'm calling ... why? example: Snom phone = Danilo 1001 Nortel phone = Marco 2002 If I call from Nortel to Asterisk , I have the display of the Snom Marco 2002 and the display of Nortel Danilo 1001 ; If I call from Nortel to Asterisk , I have the display of the Snom Marco 2002 and the display of Nortel 1001 Try placing a Wait(1) in the dialplan for the Nortel to Asterisk direction. Many Q.SIG implementations send the name in a separate message AFTER sending the SETUP message. Asterisk usually puts the call into dialplan when it receives the SETUP message. Waiting allows a subsequent message containing the name to arrive and be available in the dialplan for subsequent outward dials. 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 -- _ -- 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] changing ringtones to a group of phones
Hi all, I've been modifying the ringtone on a group of Snom phones like this, depending on certain dial-plan conditions: Exten = s,1,SIPAddHeader(Alert-Info: http://192.168.0.200/tel_ring01.wav) exten = s,n,Dial(SIP/mjc_officeSIP/mjc_homeSIP/mjc_labSIP/mjc_server,20,trj) Now, I'm migrating slowly to Digium D70 phones, which have a different Alert-Info syntax (and different ringtone names). How can I dial a group of phones simultaneously, say half Snom and half Digium, with different sip alert-info headers? - Mike -- _ -- 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] changing ringtones to a group of phones
Maybe using a LOCAL channel could help. One ext. for Snom with Snom header, another for Digium with Digium header, then simultaneously call both local channels, which then call the appropriate phones. jg -- _ -- 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] changing ringtones to a group of phones
On 05/03/2013 01:22 PM, jg wrote: Maybe using a LOCAL channel could help. One ext. for Snom with Snom header, another for Digium with Digium header, then simultaneously call both local channels, which then call the appropriate phones. Thanks, that might work! - Mike -- _ -- 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] Digium D70 visual voicemail - won't play
Hi all, I'm trying out a Digium D70 phone with Asterisk 11. My voicemail messages are listed in the visual voicemail app on the phone, but they do not successfully play back. The correct duration is shown, but the progress bar just jumps back to zero when I press the Play softbutton. I can hear my messages fine if I manually dial into my voicemail extension. I have format=wav49 in voicemail.conf. Is that a problem format for the D70? - Mike D70 Current Firmware Version: 1_3_0_2_54153 Asterisk 11.3.0 -- _ -- 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 QSIG doesnt send the calling name to Nortel CS1000
You did not show how the Nortel side is configured, especially LD 17 ADAN configuration. Regards Hans On 2013-05-03 11:27, Danilo Dionisi wrote: I'm sorry, the mail is automatically send :p However, I am for the Asterisk, there are other external consultants for Nortel ... according to you can be out a patch for Asterisk to send the facility of CPND??? Danilo Il giorno venerdì 3 maggio 2013, Danilo Dionisi dionisi.dan...@gmail.com mailto:dionisi.dan...@gmail.com ha scritto: Thanks Dale for your answer. I am a consultant for a project for Banca D'Italia and we have to give birth to 20 branches with 1,200 snom phones ... oh my god I can not record 1200 names on Nortel!!! : '( Il giorno giovedì 2 maggio 2013, Dale Noll dn...@wi.rr.com mailto:dn...@wi.rr.com ha scritto: I retired my Nortel switch a couple of years ago, but I do not believe I ever got Asterisk - Nortel to pass the CPND, just the number. If I remember correctly, I had to enter then names manually in Nortel (LD 95?) for display on the Nortel endpoints. On Tue, Apr 30, 2013 at 11:30 AM, Danilo Dionisi dionisi.dan...@gmail.com mailto:dionisi.dan...@gmail.com wrote: Hello to all, I have a problem with an asterisk qsig. I have three machines: Nortel CS1000 --- Card Sangoma PRI --- Asterisk QSIG ---SIP Trunk--- Asterisk I use Snom phones on Asterisk. If I call from Asterisk to Nortel, Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone, but if I call from Nortel to Asterisk, the QSIG does not send Nortel on the display of the name of the person i'm calling ... why? example: Snom phone = Danilo 1001 Nortel phone = Marco 2002 If I call from Nortel to Asterisk, I have the display of the Snom Marco 2002 and the display of Nortel Danilo 1001; If I call from Nortel to Asterisk, I have the display of the Snom Marco 2002 and the display of Nortel 1001 This is my / etc / asterisk / chan_dahdi.conf [channels] cc_offer_timer=20 ccbs_available_timer=4800 ccnr_available_timer=7200 cc_recall_timer=20 cc_agent_policy=native cc_monitor_policy=native pridialplan=private prilocaldialplan=private context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no facilityenable=yes callerid=asreceived ;Sangoma A104 port 1 [slot:4 bus:17 span:1] wanpipe1 switchtype=qsig context=from_nortel group=0 echocancel=yes faxdetect=incoming signalling=pri_cpe channel =1-15,17-31 -- _ -- 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
