Re: [Asterisk-Users] ISDN DID
Hi, this SETUP message does not contain a CalledParty IE. That means your telco does not send you the DID. You will probably get ripped off extra for that feature by your telco. best regards Klaus -- Klaus-Peter Junghanns On Tue, 2005-08-09 at 17:20 -0500, Panitaxx wrote: Hi, thanks for your response. here is the log of one call: Enabled debugging on span 1 Asterisk*CLI Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 72/0x48) (Originator) Message type: SETUP (5) [a1] Sending Complete (len= 1) [04 03 90 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 8d] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 13 ] [1e 02 84 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0b 00 83 39 31 35 34 35 31 39 30 30] Calling Number (len=13) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation allowed of network provided number (3) '915451900' ] -- Making new call for cr 72 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, C alling Party Number) -- Going to extension s|1 because of Complete received Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 8d] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 13 ] -- Accepting call from '915451900' to 's' on channel 0/13, span 1 Asterisk*CLI -- Executing Playback(Zap/13-1, vm-intro|noanswer) in new stack Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: PROGRESS (3) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Playing 'vm-intro' (language 'es') Asterisk*CLI -- Executing Playback(Zap/13-1, vm-goodbye) in new stack Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: CONNECT (7) [18 03 a9 83 8d] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 13 ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Playing 'vm-goodbye' (language 'es') Asterisk*CLI Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 72/0x48) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Executing NoOp(Zap/13-1, ) in new stack -- Executing Hangup(Zap/13-1, ) in new stack == Spawn extension (primario, s, 4) exited non-zero on 'Zap/13-1' NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32840/0x8048) (Terminator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/13-1' Asterisk*CLI On 8/9/05, jj [EMAIL PROTECTED] wrote: What does pri debug span 1 show? On Aug 9, 2005, at 5:02 PM, Panitaxx wrote: Hello, I have an ISDN PRI E1. For some reason I am not receiving the did number so every call can only go to s exten. I have tried using _X. exten. Also I have immediate=no in zapata.conf. Any hint? thanks in advance, Iván Aponte
Re: [Asterisk-Users] SNOM Hint for MeetMe
Hi, take a look at app_devstate. It lets you control SNOM LEDs from the dialplan, e.g.: exten = 1234,hint,DS/1234 exten = 1234,1,DevState(1234,2) ; == solid , or 1234,6 for blinking exten = 1234,2,Meetme(1234) exten = 1234,3,Hangup exten = h,1,DevState(1234,0) ; LED off The confiugre one SNOM funtion key as a destination to 1234. have fun, Klaus -- Klaus-Peter Junghanns On Mon, 2005-08-08 at 21:55 -0400, Dustin Wildes wrote: Has anyone written a php/perl or a hack to the 'hint' function in Asterisk that will let you monitor a MeetMe conference? So if anyone was in a conference, I could have a button light up on my Snom 360? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM Hint for MeetMe
There is a bristuff for CVS HEAD (quite old though...), but a newer version is on its way. On Tue, 2005-08-09 at 08:16 -0400, Dustin Wildes wrote: This would be absolutely perfect! I found the app_devstate.so in the 'bristuff' package. Has anyone ported over the app_devstate.c to work with HEAD? Or do you have to use this with bristuff's patched version of asterisk? Klaus-Peter Junghanns wrote: Hi, take a look at app_devstate. It lets you control SNOM LEDs from the dialplan, e.g.: exten = 1234,hint,DS/1234 exten = 1234,1,DevState(1234,2) ; == solid , or 1234,6 for blinking exten = 1234,2,Meetme(1234) exten = 1234,3,Hangup exten = h,1,DevState(1234,0) ; LED off The confiugre one SNOM funtion key as a destination to 1234. have fun, Klaus -- Klaus-Peter Junghanns On Mon, 2005-08-08 at 21:55 -0400, Dustin Wildes wrote: Has anyone written a php/perl or a hack to the 'hint' function in Asterisk that will let you monitor a MeetMe conference? So if anyone was in a conference, I could have a button light up on my Snom 360? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM Hint for MeetMe
hmm..extracting it from: http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC8f-CVS.tar.gz shouldnt be rocket science. ;-) good luck, Klaus On Tue, 2005-08-09 at 09:36 -0400, Dustin Wildes wrote: I had noticed the 'devicestate.c' in HEAD and was looking over both the custom-bristuff version and the HEAD to see how involved it would be. Not to be pushy or anything, but do you have an ETA of the new version? I have a client that I can get off my back if I make some of their buttons light-up! (not extensions - but settings related to astdb) *hahah* I'll be more than happy to test it out. Thanks for your help!! --Dustin Klaus-Peter Junghanns wrote: There is a bristuff for CVS HEAD (quite old though...), but a newer version is on its way. On Tue, 2005-08-09 at 08:16 -0400, Dustin Wildes wrote: This would be absolutely perfect! I found the app_devstate.so in the 'bristuff' package. Has anyone ported over the app_devstate.c to work with HEAD? Or do you have to use this with bristuff's patched version of asterisk? Klaus-Peter Junghanns wrote: Hi, take a look at app_devstate. It lets you control SNOM LEDs from the dialplan, e.g.: exten = 1234,hint,DS/1234 exten = 1234,1,DevState(1234,2) ; == solid , or 1234,6 for blinking exten = 1234,2,Meetme(1234) exten = 1234,3,Hangup exten = h,1,DevState(1234,0) ; LED off The confiugre one SNOM funtion key as a destination to 1234. have fun, Klaus -- Klaus-Peter Junghanns On Mon, 2005-08-08 at 21:55 -0400, Dustin Wildes wrote: Has anyone written a php/perl or a hack to the 'hint' function in Asterisk that will let you monitor a MeetMe conference? So if anyone was in a conference, I could have a button light up on my Snom 360? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk E1 in europe
i guess so On Wed, 2005-07-13 at 07:39 -0700, Matt wrote: is euroisdn DSS1 protocol working with asterisk? Best Regards Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quadBRI form junghanns.net
howdy, the problems with data and fax calls were mainly caused by asterisk, e.g. echo cancelation always on, failed native bridging, gains, Since bristuff 0.2.0-RC8e those issues have been solved. We have quite a few customers running loads of ISDN data calls between their locations without any special asterisk options. best regards Klaus -- Klaus-Peter Junghanns Am Donnerstag, den 07.07.2005, 09:13 +0200 schrieb Ivan Meic (Vox Mundi): I had quite a lot of experience with it ... it works fine, the only problem I got was that I couldn't transmit fax (data) calls through it reliably ... although this was some time ago, so it is possible that the kernel modules for them improved lately. Ivan Hello, Is anybody there using quadBRI form Junghanns.net with Asterisk ? I would like to order that card but first would like to hear some opinions. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quadBRI form junghanns.net
Ivan, as long as you use BRIstuff it will work fine with any zaptel hardware, even with Digium or Sangoma. best regards Klaus -- Klaus-Peter Junghanns Am Donnerstag, den 07.07.2005, 12:25 +0200 schrieb Ivan Meic (Vox Mundi): Klaus, Can the data transmission work reliably now between an incoming PRI line (Digium TE405P) and outgoing BRI line (QuadBRI) ? Ivan the problems with data and fax calls were mainly caused by asterisk, e.g. echo cancelation always on, failed native bridging, gains, Since bristuff 0.2.0-RC8e those issues have been solved. We have quite a few customers running loads of ISDN data calls between their locations without any special asterisk options. best regards Klaus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to get *8 call pickup to work
Hi, app_pickup, app_pickupchan, app_pickdown, app_steal are your friend in BRIstuff. ;) best regards Klaus Am Mittwoch, den 29.06.2005, 10:09 -0500 schrieb Brian West: Go get app_intercept from www.pbxfreeware.org /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jun 29, 2005, at 9:16 AM, Tony Nichols wrote: I have been unable to get it to pickup sip-sip calls but if an incoming zap rings I can hit *8# and it works. My config is the same as yours: zapata has callgroup = 1 and in sip.conf I have pickupgroup = 1 I'm also using Grandstreams. t o n y On 6/28/05, Robert Woodcock [EMAIL PROTECTED] wrote: I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When I call from a zap channel or a SIP phone to another SIP phone, then dial *8 from a third SIP phone, I get 503 Service Unavailable on the third phone and I get this at the Asterisk console: Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible... Jun 28 09:01:24 NOTICE[16774]: chan_sip.c:7402 handle_request: Nothing to pick up I'd appreciate hearing from anyone that has this working. Here's my sip.conf, features.conf, and zapata.conf: # zapata.conf sed 's/;.*//g' | grep -v ^$ [trunkgroups] [channels] context=default switchtype=national signalling=em_w rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived callprogress=yes musiconhold=default channel = 1-24 # features.conf sed 's/;.*//g' | grep -v ^$ [general] parkext = 700 parkpos = 701-720 context = parkedcalls pickupexten = *8 # sip.conf sed 's/;.*//g' | grep -v ^$ | grep -v '^[ ]' | sed s/ secret=.*/secret=donttell/g [general] context=default callgroup=1 pickupgroup=1 port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=alaw allow=g723.1 allow=g729 callgroup=1 pickupgroup=1 context=default nat=no canreinvite=yes dtmfmode=rfc2833 incominglimit=4 [1310] username=1310 secret=donttell type=friend host=dynamic callerid=Grandstream SIP 1310 [EMAIL PROTECTED] [i1310] username=i1310 secret=donttell type=friend host=dynamic callerid=Grandstream SIP 1310 [1311] username=1311 secret=donttell type=friend host=dynamic callerid=John Jacob Jingleheime 1311 [EMAIL PROTECTED] [1312] username=1312 secret=donttell type=friend host=dynamic callerid=Cisco 7960G Test 1312 [EMAIL PROTECTED] FWIW, I get identical behavior with callgroup=/pickupgroup= specified for each extension. Here's some sanitized verbose output with SIP debugging enabled: -- Starting simple switch on 'Zap/24-1' Jun 28 10:43:18 DEBUG[16774]: chan_sip.c:771 __sip_autodestruct: Auto destroying call 'a01052a-13c4-42c104ea-371e-1957' Destroying call 'a01052a-13c4-42c104ea-371e-1957' Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 1 on Zap/24-1 Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 3 on Zap/24-1 Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 1 on Zap/24-1 Jun 28 10:43:20 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 2 on Zap/24-1 Jun 28 10:43:20 DEBUG[17450]: chan_zap.c:1381 zt_enable_ec: Enabled echo cancellation on channel 24 -- Executing Macro(Zap/24-1, stdexten|1312|SIP/1312) in new stack -- Executing Dial(Zap/24-1, SIP/1312|20) in new stack Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1309 create_addr: Setting NAT on RTP to 0 Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1487 sip_call: Outgoing Call for 1312 Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1620 update_user_counter: Call from user '1312' is 1 out of 0 We're at asterisk.server.ip.addr port 19630 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x1 (g723) Answering with preferred capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 13 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0 Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760 From: asterisk sip:[EMAIL PROTECTED];tag=as61d8a13d To: sip:[EMAIL PROTECTED]:5061 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 28 Jun 2005 17:43:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 284 v=0 o=root 17450 17450 IN IP4 asterisk.server.ip.addr s=session c=IN
Re: [Asterisk-Users] cheap HFC card on Bristuff vs cheap HFC card on i4l vs Fritz ISDN BRI card on CAPI
Howdy, Am Dienstag, den 28.06.2005, 09:01 +0200 schrieb vdasilva: Hello I have asterisk running in Red Hat 9 with a cheap HFC card on i4l. I have choppy sound problems sometimes, and echo problems often. I am using a 2 port Grandstream ATA, Grandstream BT and a Grandstream GPX-2000 I read that changing to BriStuff will fix the echo problems, but have also read other users say that the only way they solved the echo/choppy sound problems was using a Fritz ISDN card with the CAPI drivers... Yes, BRIstuff and the hfc-pci will provide echo cancelation. With the Fritz card however you will NOT get echo cacnelation. I have tried using bristuff on RH9 but couldn't get my zaptel to compile... Do you have _configured_ kernel sources installed? If you run a 2.6 kernel do you have the necessary scripts to build kernel modules (these are built during the kernel compilation process)? Then there is the issue of timing, ztdummy or zaprtcand QoS setup on the Linux box... Can anyone who has a 100% working Asterisk implementation using any of the techniques described above tell me more... I will happily upgrade to the Fritz card if it will solve all the problems... Thanks Vicente best regards Klaus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junghanns 4 port BRI problem
Hi, CRC errors are caused by bit errors on layer 1. In most cases this is a cable issue. Did you try replacing the cable from the NT1 to the quadBRI? How long is that cable? However if only 1 of the 2 B channels are working then you might have your BRI lines get checked or try a different ISDN device on those lines. best regards Klaus -- Klaus-Peter Junghanns Am Dienstag, den 28.06.2005, 18:22 +0200 schrieb Doug Reid - Stormcorp: Hi All I have a Junghanns BRI 4 port installed where only the first channel of each line is working i.e. channels 1 and 4 work but 2 and 5 don't. Our config is the same on this box as 15 other similar installations where all works well. the only error I see is in /var/log/messages: Jun 28 15:49:31 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:51:27 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:53:09 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:56:48 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:58:06 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 16:01:01 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Can anyone help with this? Thanks Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Correction to Janghanns BRI problem
Hi, what signalling does the telco run on those lines? best regards Klaus Am Dienstag, den 28.06.2005, 19:02 +0200 schrieb Doug Reid - Stormcorp: Hi all Correction on my last mail, I found that line 1 both channels work but on line 2 none work. I have 2 BRI ISDN lines coming in on port 1 and 2 (4 channels) on a Junghanns 4 port. The setup by the Telco on this ISDN is different than our others, they have 2 lines (4 channels) that are all connected to one telephone number i.e. 701 5161. The second number should be 701 5162 but this number does not exist. If we put a Sirrix card in all 4 channels (2 x BRI) work fine on 701 5161 but when we put a Junghanns in only one line works. It seems like the second line is not given a B channel from the NTU side of the Telco. Error in /var/log/messages: Jun 28 15:49:31 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:51:27 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:53:09 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jun 28 15:56:48 pbxct kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Please if anyone could suggest a fix here it would be much appreciated. Thanks Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff-0.2.0-RC8h does not compile
Hi, it helps to have configured and working kernel sources installed. Configure your kernel sources for the running kernel and then run make in the kernel source dir to build the necessary scripts. You dont have to wait until the kernel is compiled. best regards Klaus -- Klaus-Peter Junghanns Am Samstag, den 25.06.2005, 02:45 +0200 schrieb Stefan Gofferje: Hi folks, I just tried to compile the latest bristuffed asterisk on a SuSE 9.2 Pro but the compilation stopped with errors. Anyone any comments on that? rm -f qozap.o *.ko *.mod.c *.mod.o .*o.cmd *~ rm -rf .tmp_versions make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/bristuff-0.2.0-RC8h/qozap ZAP=-I/usr/src/bristuff-0.2.0-RC8h/zaptel-1.0.8 modules make[1]: Entering directory `/usr/src/linux-2.6.8-24.14' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.8-24.14' make: *** [linux26] Error 2 install -D -m 644 qozap.ko /lib/modules/`uname -r`/misc/qozap.ko install: cannot stat `qozap.ko': No such file or directory make: *** [installlinux26] Error 1 quadBRI driver installed. Press Enter to continue, or CTRL + C to abort. rm -f cwain.o *.ko *.mod.c *.mod.o .*o.cmd *~ make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/bristuff-0.2.0-RC8h/cwain ZAP=-I/usr/src/bristuff-0.2.0-RC8h/zaptel-1.0.8 modules make[1]: Entering directory `/usr/src/linux-2.6.8-24.14' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.8-24.14' make: *** [linux26] Error 2 install -D -m 644 cwain.ko /lib/modules/`uname -r`/misc/cwain.ko install: cannot stat `cwain.ko': No such file or directory make: *** [installlinux26] Error 1 cwain driver installed. Press Enter to continue, or CTRL + C to abort. rm -f zaphfc.o *.ko *.mod.c *.mod.o .*o.cmd *~ rm -rf .tmp_versions make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/bristuff-0.2.0-RC8h/zaphfc ZAP=-I/usr/src/bristuff-0.2.0-RC8h/zaptel-1.0.8 modules make[1]: Entering directory `/usr/src/linux-2.6.8-24.14' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.8-24.14' make: *** [linux26] Error 2 install -D -m 644 zaphfc.ko /lib/modules/`uname -r`/misc/zaphfc.ko install: cannot stat `zaphfc.ko': No such file or directory make: *** [installlinux26] Error 1 hfc-pci driver installed. Press Enter to continue, or CTRL + C to abort. Regards, Stefan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] isdn channels busy
Hi, that is a bug in libpri. You will sometimes notice a message like: !! No channel map, no channel, and no ds1? What am I supposed to identify? This is caused by a restart message from the switch containing no channel ident IE. According to the ETSI standard this indicates a restart of all B channels on that span. Libpri just ignores this message. Now all B channels are blocked for incoming calls. You can free them by making outgoing calls. Fixed in BRIstuff since May. ;) best regards Klaus -- Klaus-Peter Junghanns Am Samstag, den 25.06.2005, 13:47 +0100 schrieb Asterisk: We've got a EuroISDN (32 channels) with a TE405p, running cvs head as of 5 days ago. In the past couple of days, we've hit a scenario where incoming calls to the * pbx from the PSTN are being marked as busy, but outgoing calls work just fine. When we reboot *, the problem goes away. Has anyone else had this ? I've attached a PRI debug below. I've changed the phone numbers (x y) to protect the innocent :) Please tell me that there is someone who has had this issue, and knows how to get round it. It's making my users, well, irate ... Many thanks. Julian pbx*CLI [Span 2 D-Channel 0] Protocol Discriminator: Q.931 (8) len=43 [Span 2 D-Channel 0] Call Ref: len= 2 (reference 1/0x1) (Originator) [Span 2 D-Channel 0] Message type: SETUP (5) [a1] [Span 2 D-Channel 0] Sending Complete (len= 1) [04 03 90 90 a3] [Span 2 D-Channel 0] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) [Span 2 D-Channel 0] Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) [Span 2 D-Channel 0] Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] [Span 2 D-Channel 0] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 [Span 2 D-Channel 0]ChanSel: Reserved [Span 2 D-Channel 0] Ext: 1 Coding: 0 Number Specified Channel Type: 3 [Span 2 D-Channel 0] Ext: 1 Channel: 1 ] [1e 02 84 83] [Span 2 D-Channel 0] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) [Span 2 D-Channel 0] Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0c 21 83 3x 3x 3x 3x 3x 3x 3x 3x 3x 3x] [Span 2 D-Channel 0] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) [Span 2 D-Channel 0] Presentation: Presentation allowed of network provided number (3) 'xx' ] [70 07 81 34 34 34 37 30 35] [Span 2 D-Channel 0] Called Number (len= 9) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'yy' ] [Span 2 D-Channel 0]-- Making new call for cr 1 [Span 2 D-Channel 0]-- Processing Q.931 Call Setup [Span 2 D-Channel 0]-- Processing IE 161 (cs0, Sending Complete) [Span 2 D-Channel 0]-- Processing IE 4 (cs0, Bearer Capability) [Span 2 D-Channel 0]-- Processing IE 24 (cs0, Channel Identification) [Span 2 D-Channel 0]-- Processing IE 30 (cs0, Progress Indicator) [Span 2 D-Channel 0]-- Processing IE 108 (cs0, Calling Party Number) [Span 2 D-Channel 0]-- Processing IE 112 (cs0, Called Party Number) [Span 2 D-Channel 0]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Present, peerstate Call Initiated [Span 2 D-Channel 0] Protocol Discriminator: Q.931 (8) len=9 [Span 2 D-Channel 0] Call Ref: len= 2 (reference 1/0x1) (Terminator) [Span 2 D-Channel 0] Message type: RELEASE COMPLETE (90) [08 02 81 ac] [Span 2 D-Channel 0] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) [Span 2 D-Channel 0] Ext: 1 Cause: Requested channel not available (44), class = Network Congestion (2) ] [Span 2 D-Channel 0]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null [Span 2 D-Channel 0]NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null pbx*CLI ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HDLC abort 6 error
Hi, a common reason for HDLC aborts is interrupt latency/jitter. Most likely when you are sharing an IRQ, are not using DMA mode for IDE disks or your IDE controller is disableing all IRQs while it is servicing his own. If you have an IDE system please check: hdparm -d /dev/hdX hdparm -u /dev/hdX Welcome to the worls of software hdlc. :-) best regards Klaus -- Klaus-Peter Junghanns Am Donnerstag, den 23.06.2005, 20:59 -0500 schrieb [EMAIL PROTECTED]: I've read as much as I can on this error and still can't seem to figure out what's causing this error: Jun 23 20:54:58 NOTICE[7483]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 23 20:55:03 NOTICE[7483]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Jun 23 20:55:03 NOTICE[7483]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 If I restart my entire machine it will work and assign the D channel correctly , but after a few minutes it then starts producing this error. I have changed the second number in my /etc/zaptel.conf span='s line to a 0, as instructed by Digium and still have the same issue. I'm running Gentoo with plain old 2.6 vanilla sources. My Te110p card is the only thing on it's irq and when I run a zttest I get all 100% - 99.975586 (lowest). I'm really at a loss here. My card has a solid green light. It's a Bellsouth PRI 12 -bchannel and 1 dchannel new install. Any ideas, please help. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRIstuff/QuadBRI problem: Ring requested on unconfigured channel 255/255 span 5
Hi, can you please post the output of zap show channel for all channels of an affected span? It seems that asterisk thinks that all B channels are still in use. So i suspect some problem with call clearing. best regards Klaus -- Klaus-Peter Junghanns Am Freitag, den 24.06.2005, 12:07 +0200 schrieb [EMAIL PROTECTED]: Hi all, I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns QuadBRI running in NT-mode. Connected to the BRI-Ports are 12 Fax-Modems (Elsa MicroLink ISDN/TL V.34) which are only operating in dial out analog mode to deliver fax messages. After a while of running fine (50-200 dial out connections) on some S0 spans the following message occurs over and over again: chan_zap.c:8009 pri_dchannel: Ring requested on unconfigured channel 255/255 span 5 The Modems connected to this span get NO DIALTONE for every ATD. Modems on other spans continue to operate. This error seems to appear mostly on span 5 and 6. After restarting asterisk everything is okay again for a while. Any hints about what is going on here are greatly appreciated ;-) TIA, Bruno - bruno @ ic3s.de Connected to Asterisk 1.0.7-BRIstuffed-0.2.0-RC8g currently running on pbx (pid = 20305) Verbosity is at least 5 pbx*CLI show channels Channel (ContextExtensionPri ) State Appl. Data Zap/25-1 (pri1 s1 ) Up Bridged Call Zap/128-1 Zap/128-1 (from-s0-faxmodems 00711xxx 5 ) Up Dial Zap/r1/0711xxx Zap/24-1 (pri1 s1 ) Up Bridged Call Zap/132-1 Zap/132-1 (from-s0-faxmodems 00242xxx 5 ) Up Dial Zap/r1/0242xxx 4 active channel(s) Jun 24 11:49:19 WARNING[20305]: chan_zap.c:8009 pri_dchannel: Ring requested on unconfigured channel 255/255 span 5 == Primary D-Channel on span 6 down for TEI 64 == Primary D-Channel on span 6 up for TEI 64 -- Accepting overlap voice call from '' to '00394' on channel 0/2, span 6 -- Starting simple switch on 'Zap/129-1' -- Channel 0/2, span 7 got hangup -- Hungup 'Zap/24-1' == Spawn extension (from-s0-faxmodems, 00242xxx, 5) exited non-zero on 'Zap/132-1' -- Hungup 'Zap/132-1' -- Executing SetCallerPres(Zap/129-1, prohib) in new stack -- Executing NoOp(Zap/129-1, ) in new stack -- Executing SetTransferCapability(Zap/129-1, 3K1AUDIO) in new stack -- Setting transfer capability to: 0x10 - 3K1AUDIO. -- Executing SetCIDNum(Zap/129-1, 0410612345) in new stack -- Executing Dial(Zap/129-1, Zap/r1/039) in new stack -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called r1/039 -- Zap/26-1 is ringing -- Zap/26-1 answered Zap/129-1 -- Attempting native bridge of Zap/129-1 and Zap/26-1 Jun 24 11:49:35 WARNING[20305]: chan_zap.c:8009 pri_dchannel: Ring requested on unconfigured channel 255/255 span 5 == Primary D-Channel on span 7 down for TEI 65 == Primary D-Channel on span 7 up for TEI 65 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement
Yes, that should be possible. But I don't think a channel driver (and each channel driver) should do that on its own. Software echo cancelling belongs in a common part of Asterisk. I strongly agree. But asterisk doesn't seem to work this way. Zap channel has it's own echo cancel engine. Other channels don't. This is so sad :-( Why not implement a really common echo cancel api usable from any channel ?? Exactly! I'm not familiar with the Asterisk API, but it could be some plugin like res_* ... Maybe this belongs to the Asterisk-Dev list. Armin I strongly disagree. :-) You dont want to do echo cancelation in userspace. Especially not on a non-realtime operating system. To make echo cancelation work it has to be as close to the line interface as possible. Also the frames have to be as small as possible. This rules out capi pretty much. best regards Klaus -- Klaus-Peter Junghanns ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement
Am Donnerstag, den 23.06.2005, 12:41 +0200 schrieb Armin Schindler: I strongly disagree. :-) You dont want to do echo cancelation in userspace. Especially not on a non-realtime operating system. To make echo cancelation work it has to be as close to the line interface as possible. Also the frames have to be as small as possible. This rules out capi pretty much. If you don't want echo-canceling in user-space, then neither Asterisk nor any chan_* plugin should do it. I don't know the zap channel code, but does the zap echo-cancel-code is inside a kernel module? Yes, sir. If yes, then I have to disagree here. Something like 'playing' with audio-data is nothing the kernel should be concerned with. This belongs in user-space and if you need realtime, then you should use a realtime OS or use RT-scheduling. Just putting such a code into kernelspace is a bad idea. What is so bad about playing with audio-data in kernel space? If you play with echo cancelation in user space you will need to de-jitter the audio first introducing more and more latency, so your echo cancelation becomes way more computationally expensive. So the correct way is either the hardware supports it or the application knows what to do with the data received, like DTMF. Why should the application have to worry about things like echo cancelation? Zaptel is not only used by Asterisk but also by other projects. With EC in kernel space (which gets switched on and off by userspace) there is no need to reinvent the EC-wheel for every project. Klaus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi-cm-0.5 release announcement
If yes, then I have to disagree here. Something like 'playing' with audio-data is nothing the kernel should be concerned with. This belongs in user-space and if you need realtime, then you should use a realtime OS or use RT-scheduling. Just putting such a code into kernelspace is a bad idea. What is so bad about playing with audio-data in kernel space? Besides preemption or RT-patches, it is not easy (and noboady does it) to be 'nice' and have a fair scheduling. With such work in kernel, you just say I'm at the highest priority, I don't care about others. And that's just wrong in the kernel. That is actually what you want to do if your system is a PBX. You want to give as much as priority to your audio quality as you can. Even if this means that userspace applications get unfair scheduling results. If you take care of the critical audio handling (like EC) inside the kernel then your (maybe very unexperienced) users cannot easily disturb this process by causing a high load in user space, e.g. by running webservers, fileservers, mailservers or X on their PBX! It's far better to have good audio quality (with a working EC) and a slowed down webserver than a garbled audio and fast webserver. Just my 2 eurocents. Normaly, the kernel just should provide access to the hardware and basic functions for interaction with the hardware. If you play with echo cancelation in user space you will need to de-jitter the audio first introducing more and more latency, so your echo cancelation becomes way more computationally expensive. That depends on what scheduling priority this task runs. If you don't want to get interrupted by other tasks, then you need a higher priority. This is true in a perfect world. :) However there are lots of nasty things in your linux box (like harddisk controllers hogging your cpu, etc...) that make your system a non-realtime system. So the correct way is either the hardware supports it or the application knows what to do with the data received, like DTMF. Why should the application have to worry about things like echo cancelation? In the case of Asterisk and echo-cancel, this application is the position where is makes sense. Otherwise you need a copy of the echo-cancel code in each channel driver. Zaptel is not only used by Asterisk but also by other projects. With EC in kernel space (which gets switched on and off by userspace) there is no need to reinvent the EC-wheel for every project. Okay, from that point of view it makes sense. On the other hand, something like echo-cancel and DTMF is not channel specific and therefore should not be part of that. Or would you add additional codecs into the channel driver? I would even put more things into kernel space just to reduce latency. There are people that would even enjoy RTP in kernel space. Running things in userspace makes sense from a software architectural point of view. But in real life this can be very dangerous if you dont have control over the complete userspace (e.g. users on crack running make bzImage -j). Armin Klaus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call inband progress indication and zaphfc
Hi, priindication = passthrough in zapata.conf is your friend. :) You need to BRIstuff your * though... best regards Klaus Am Freitag, den 10.06.2005, 12:17 +0200 schrieb Diego Ercolani: Hello all, I've a little clue with zaphfc used to connect to a BRI linethat probably can be a configuration issue (really I hope so) Here, telcos (expecially mobile operators) use to substitute the dialtone with some vocal indication without answer the line. (Indications like The customer is not reachable or wait because the customer is on the phone ecc..) For asterisk this condition is a normal dial tone and the message from the telco and it's not possible to listen theese indications. As I'm using zaphfc and with X100p and a normal analog line I can listen these indications, my question is Have you tryed with PRI cards? as I don't know if this is an issue of asterisk, zaphfc or my configuration. Thank you in advance Diego ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] send and receive MMS
Hi, I assume that you are talking about fixed line MMS like it is implemented in Germany. Some time ago i already played a little bit with a Gigaset SL74 (and an ISDN dect base). So far as ISDN is concerned the basestation uses a PPP connection to connect to a HTTP Server for sendind/retrieving MMSes. With a little effort and time this can easily be emulated by using ZapRAS/PPPd and some PHP/Perl scripts. I am not sure but i think that incoming MMSes are signalled by sending a SMS with some info and an URL. With app_SMS and some scripts you could automatically retrieve the MMS then and forward it as an Email (or another MMS to a Gigaset...). best regards Klaus -- Klaus-Peter Junghanns Am Mittwoch, den 01.06.2005, 11:54 +0200 schrieb Yannick Daronnat: Hello, did anyone already experience MMS? SMS works fine, but I can't find infos on how to send and receive MMS on a similar way with Asterisk. Thanks Daryan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does ISDN really work?
Hi, you will need app_settransfercapability to make this work properly. This is part of CVS-HEAD. I have backported it for the asterisk stable version of bristuff (see www.junghanns.net/asterisk/) and also fixed some bugs in Asterisk that will make ISDN data calls unreliable (or in some cases impossible). On the * CLI do a show application settransfercapability to find out the correct arguments. best regards Klaus -- Klaus-Peter Junghanns Am Dienstag, den 31.05.2005, 11:54 +0200 schrieb Daniel Nystrm: I'm trying to setup DATA calls with Dial(Zap/g1d/12345678), but with PRI DEBUG SPAN 1 on, it seems to connect a regular SPEECH call. I'm using 1.0.6. Is this feature broken in stable release? There seems to be support in the source, but it doesn't work. Does the Telco set what each PRI channel support? Like DATA or SPEECH etc.. Do I have to specify in zapata.conf or zaptel.conf that the channels are DATA capabale? Please help! This is driving me crazy soon. :) -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Early B3 connects on zaphfc
Hi, zapata.conf: prindication = passthrough best regards Klaus -- Klaus-Peter Junghanns Am Dienstag, den 24.05.2005, 10:34 +0200 schrieb Steven Lam: Hi, Is there a way to hear what your telco has to say (Early B3 connects) using zaphfc (zaptel)? All suggestions are welcome ;-) Gr. Steven ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN data connection through Asterisk
Howdy, Am Montag, den 23.05.2005, 18:41 +0200 schrieb Marcin: Torsten Krueger wrote: On Sat, 21 May 2005, Marcin wrote: Is there a simply way to allow dialout from ISDN modem to outside number through Asterisk? We've done this several times with Junghanns Cards - nearly no problem, just the normal dialplan entries. Thanks a lot. I'm amazed it's so easy. The only mentionable thing is, that we had to take away all settings for txgain and rxgain in zapata.conf for the affected channels. Update to 0.2.0-RC8e and all your gain problems will be solved. This release also fixes problems with data calls if you have the t or T app_dial options. best regards Klaus -- Klaus-Peter Junghanns ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 PRI Warnings
Am Montag, den 23.05.2005, 12:07 -0400 schrieb Jorge Verastegui:
Hi,
I've connected a TE110P from digium with a 2 E1 to a Siemens PBX using
asterisk from ubuntu linux.
Everything is working as expected. This box is being used as a H323
gateway to the pstn. There are few complains but it is working pretty
well overall.
There is one thing that is bothering me.
Asterisk says:
May 22 05:03:39 WARNING[9360]: PRI: !! No channel map, no channel, and no
ds1? What am I supposed to identify?
May 22 05:03:39 WARNING[9360]: PRI: !! Unable to add IE 'Channel
Identification'
That is a bug in libpri. The Siemens sends a RESTART message containing
no Channel Identification IE. This is perfectly valid according to the
ETSI standard. It is supposed to restart the entire interface ( == all
B channels). Asterisk doenst reply to this RESTART message with a
RESTART ACKNOWLEDGE message so the Siemens might be unsure about the
state of the B channels and wont send incoming calls to Asterisk
(however outgoing calls might still work and restart B channels
on those used channels).
A simple fix (ripped from bristuff) would be to put a:
if (msgtype == Q931_RESTART_ACKNOWLEDGE) {
return 0;
}
before the lines:
pri_error(!! No channel map, no channel, and no ds1? What am I
supposed to identify?\n);
return -1;
in q931.c (function static FUNC_SEND(transmit_channel_id)).
Then Asterisk replies with a RESTART ACK message containing no
Channel Identification IE.
best regards
Klaus
--
Klaus-Peter Junghanns
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Re: [Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
chan_capi 0.3.5 lacks proper support for passing on isdn cause codes to Asterisk. This is already fixed in my development version and will be in 0.4.0. :) best regards Klaus Am Samstag, den 14.05.2005, 14:50 +0200 schrieb Armin Schindler: On Fri, 13 May 2005, Elmar Haneke wrote: Then I hope to receive some reports on what is buggy/not working, wishlist and hopefully also some reports on what works well. There are at least two anoying bugs: 1. The Busy-Applicatzion does not work, there seems to be no was to singnal Busy to the caller is no SIP-Phone is ready to answer the call. 2. Dial-Application does not really detect the reason for Failings. As an Example you should have a look at the LCR script available at Telefonsparbuch.de: The script trys to do some Fallback but it does not work with chan_capi. Thanks, for pointing out such issues. But can you please be more specific and give an example on how to reproduce it? For example, if you use an Point-to-Multipoint ISDN connection (not 'Anlagenanschluss'), then you won't get an immediate 'BUSY' on SIP Busy/Congestion. It's not possible to signal the caller 'Busy' or 'Reject', because there is a timeout on the ISDN-Bus for ANY OTHER device which may answer the call. Only on timeout, the Busy is signaled. So what type of connection and environment do you use? Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi, chan_misdn and chan_modem
Hi, time to clear some things up. :) The new version of chan_capi (0.4.0) is still work in progress (no, I have not dropped chan_capi in favour of BRIstuff). I harmonized the dialstring syntax with chan_zap, so you can just use CAPI/g1/... instead of those strange constructions with the outgoing msn. It also contains fixes (contributed by Jan Stocke) to make it work on BSD. Also it will then work properly with p2p BRIs in Austria. Chan_capi 0.4.0 will work with Asterisk stable and cvs head. It does not distinguish between certain card types (CAPI means Common isdn API), maybe you (or the Wiki) are mistaking this with BRIstuff. In the beginning BRIstuff was only intended as a driver package for our BRI hardware. But more and more enhancements (to asterisk and libpri) were added and i merged and maintain patches and applications from other people that were contributed under the GPL (and thus could not be integrated into the asterisk cvs tree). We provide a version for Asterisk stable and for cvs head. If you compare chan_capi, bristuff and chan_misdn then chan_capi and chan_misdn would fall into the same category as they are just channel drivers which do not touch the asterisk core at all. BRIstuff changes some things in Asterisk to better support European users and contains modificatiosn that we made for clients. Regarding stability chan_capi and BRIstuff (for Asterisk stable) will fall into the same category. They are used in hundreds of production installations around the globe. I cannot comment on the stability of chan_misdn because i have never used it (i have read the source though), but i made the experience the authors of chan_misdn (Beronet) supply patches for bristuff to their customers that enable them to use BRIstuff with their hardware (instead of chan_misdn). They do not distinguish between cards (so also our Junghanns.NET cards work with chan_misdn) because chan_misdn does not talk directly to the card. This is done by the mISDN kernel modules. The driver for the HFC-4S/8S based cards (used in the Junghanns.NET amd Beronet cards) was not written by Beronet but by the author of PBX4Linux, Andreas Eversberg. So, for your hfc-pci based isdn card you can use the zaphfc module from BRIstuff and use it with chan_zap OR you can use it with the mISDN driver and chan_misdn OR you can use it with the mISDN driver plus the capi layer of mISDN and chan_capi. I also have a W6692 card laying on my desk (contributed by Michael Sandee) and will write a zaptel driver for that card, but this is rather a longterm project. ;) best regards Klaus -- Klaus-Peter Junghanns Am Freitag, den 13.05.2005, 08:46 +0200 schrieb Jan Louw: Could someone please comment on the current state of chan_capi, chan_misdn and chan_modem channel drivers in terms of functionality (echo cancelation, fax, latency etc) and stability. Specifically, which channel driver would be best for a passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that chan_capi distinguishes between junghanns and non-junghans cards, and that chan_misdn is better suited for general misdn compatibility. A second point I'd like some clarification on is the purpose of Junghann's BRIStuff patch. Is this patch only necessary for chan_capi or also for chan_misdn? Does this patch add functionality to asterisk or is it only intended to smooth chan_capi integration into asterisk? Thanks in advance! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HFC-5/S + Asterisk
Am Mittwoch, den 02.02.2005, 10:41 +0100 schrieb Thomas Niesel: On Wed, Feb 02, 2005 at 08:06:35AM +0100, Peer Oliver Schmidt wrote: Thomas Niesel wrote: [..] = Your Card should work with i4l (bad) and zaphfc from junghanns (good) Forget about Capi and mISDN, go for kernel 2.6.10 along with zaphfc, ztdummy together with uhci for timing and ask the wiki for more details. Pardon my ignorance, but isn't one of the reasons for zaphfc to provide a ZAP timing source? So, if you have zaphfc card together with the bristuff from http://www.junghanns.net you don't need ztdummy, do you? Timing is done either by Hardware (Digium cards), zaprtc (a rewrite from junghanns for the kernel-rtc module), via rtai (new in bristuff RC5) or via uhci usb (ztdummy). HFC cards do not have timing hardware, so you need something else. HFC cards do have a timing source. If you run zaphfc you dont need any other zaptel timing source. best regards Klaus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 crashes with Ouch ... error while writing audio data: : Broken pipe
Hi, please start asterisk -vvvcg (so it creates a core file when it segfaults), then run gdb /usr/sbin/asterisk corefile, hit Enter a few times and run a backtrace using bt. Please email the output. I doubt that it's bristuff bug, since many users have already successfully upgraded. best regards Klaus Am Montag, den 31.01.2005, 08:33 +0100 schrieb Remco Barende: On Sun, 30 Jan 2005, Martin List-Petersen wrote: Citat Remco Barende [EMAIL PROTECTED]: I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything seems to be running fine but after some time asterisk just goes crazy (even withouth any incoming or outgoing call activity perviously). If I leave the box up for some time * goes haywire and the console is flooded with this message: Ouch ... error while writing audio data: : Broken pipe At that time I can see that there are multiple instances of mpg123 active. The solution to this problem is to kill-9 mpg123, do the same for *, unload the modules and then load the modules again and start asterisk. If I do not unload re-load the modules I cannot access the ISDN line nor do incoming calls work. I really don't know where to look for this problem. Is it possible to completely disable music on hold? Asterisk combined mpg123 is causing nothing but problems anyway, the current stable still leaves abandoned mpg123 processes. It doesn't work :( Asterisk doesn't go haywire flooding the console but now simply bombs out with : *CLI Segmentation fault I guess that qualifies it as a bristuff bug? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Q: PRI leading 0 (area access code) or 00 (country access code) missing on incoming callerid
Hi, standard asterisk doesnt support that. However it's in bristuff (www.junghanns.net/asterisk) zapata.conf: nationalprefix=0 internationalprefix=00 best regards Klaus Am Montag, den 31.01.2005, 20:04 +0100 schrieb Frank Sautter: hi, on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. only prepending a single 0 is not the solution as suggested by some writers on this list, because there is no way to differ between national and international callerids and it's not possible to make the decission based on the length of the presented callerid, as the length of the callerid can vary in most countries. e.g.: i'm getting signalled 4123456789 which could be a call from Barmstedt (Germany) which has the areacode '4123' or from Switzerland which has the countrycode '41' somehow our ericsson businessphone 250 fromerly connected to the same E1-PRI was capable of showing the correct number of leading 0s?!? regards frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk at CeBit 2005
Hi there too, we will be present at CeBIT 2005 too. At our booth 51D in Hall 13 we will be showing Asterisk software and hardware (mostly bristuffed). If you have questions or configuration problems please come and visit our Asterisk helpdesk. Come and take a look! :) We will also show the farfon (the worlds first IAX2-only VoIP phone www.farfon.com). best regards Klaus -- Klaus-Peter Junghanns http://www.junghanns.net Am Montag, den 31.01.2005, 23:16 +0100 schrieb Thilo Rler: Hi there, I just wanted to point out that Asterisk will be present at CeBit this year. We gathered some money from sponsors and were able to afford a booth together with a training-company. We'd be happy to find others joining us at the booth somewhere between 10th and 16th of March in Hannover, Germany :-) Kind regards ... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.0.2-BRIstuffed-0.2.0-RC2b and '*8' calls dropping
Hi Mark, please take a look at bristuff 0.2.0-RC5 which uses * 1.0.5: http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC5.tar.gz best regards Klaus Am Freitag, den 28.01.2005, 14:35 +0200 schrieb Mark Elkins: I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a call with '*8' - the call will drop after about 20 or so seconds. Is this a general problem with Asterisk 1.0.2? As this is the latest release that it appears Klaus-Peter Junghanns has for public consumption - is there anything I can patch for just this problem - or has Klaus-Peter Junghanns (or anyone else) been quietly busy with a BRIstuffed patch that works against Asterisk Head? I also notice that I can't seem to re-compile the H323 stuff any more... with this release... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Restart - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
Hi, restarting the B channels is a normal process on PRIs. Nothing to worry about as long only idle B channels are restarted. best regards Klaus Am Donnerstag, den 27.01.2005, 13:10 +0100 schrieb Frank Sautter: hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * from time to time (sometime within a few minutes sometime after hours) a complete PRI line or several PRI lines are kind of resetting (none of my colleagues reported a call interruption though). could this be a problem of the length (around 4kilometres) of the line between the telco switch and the NT providing the E1-PRI? The PRI line itself is only 3 metres long. is this the line build-out parameter in /etc/zaptel.conf? or is this something with timing of the span? my current settings are: # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) # TE405P/TE410P quad E1 span=2,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 span=3,0,0,ccs,hdb3,crc4 bchan=36-50,52-66 dchan=51 span=4,2,0,ccs,hdb3,crc4 bchan=67-81,83-97 dchan=82 span=5,0,0,ccs,hdb3,crc4 bchan=98-112,114-128 dchan=113 this is a excerpt from /var/log/asterisk/full -- B-channel 0/1 successfully restarted on span 2 -- B-channel 0/3 successfully restarted on span 2 -- B-channel 0/5 successfully restarted on span 2 -- B-channel 0/6 successfully restarted on span 2 -- B-channel 0/7 successfully restarted on span 2 -- B-channel 0/8 successfully restarted on span 2 -- B-channel 0/9 successfully restarted on span 2 -- B-channel 0/10 successfully restarted on span 2 -- B-channel 0/11 successfully restarted on span 2 -- B-channel 0/12 successfully restarted on span 2 -- B-channel 0/13 successfully restarted on span 2 -- B-channel 0/14 successfully restarted on span 2 -- B-channel 0/17 successfully restarted on span 2 -- B-channel 0/18 successfully restarted on span 2 -- B-channel 0/19 successfully restarted on span 2 -- B-channel 0/20 successfully restarted on span 2 -- B-channel 0/21 successfully restarted on span 2 -- B-channel 0/22 successfully restarted on span 2 -- B-channel 0/23 successfully restarted on span 2 -- B-channel 0/24 successfully restarted on span 2 -- B-channel 0/25 successfully restarted on span 2 -- B-channel 0/26 successfully restarted on span 2 -- B-channel 0/27 successfully restarted on span 2 -- B-channel 0/28 successfully restarted on span 2 -- B-channel 0/29 successfully restarted on span 2 -- B-channel 0/30 successfully restarted on span 2 -- B-channel 0/31 successfully restarted on span 2 regards frank sautter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bristuff ZapHFC and Loosing D-Channel
Hi, that is the usual behaviour on a P2MP BRI line. When idle the telco will bring down layer 2 and layer 1. Bristuff will activate layer 1 and layer 2 again immediately. best regards Klaus Am Donnerstag, den 27.01.2005, 16:01 +0100 schrieb Remco Barende: Hi! Did you ever find the answer to your question? I am getting the same message on the console every second: == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up etc. etc. etc. I'm running Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 The error is only visible however if I run * with -v (but I guess I shouldn't see these messages nonetheless)? On Tue, 25 Jan 2005, Peer Oliver Schmidt wrote: Using the latest(?) bristuff (Asterisk 1.0.4-BRIstuffed-0.2.0-RC3a) I have problems with loosing the D-channel. Most of the time, after the message PRI D-channel down it only takes a second or so to come back up, noted by the message PRI D-channel up However, today most of the time the D-channel stays down. Calls come in, but can't be answered. Does anyone know of a fix for this, or might have some insights on how to circumvent this problem? Any and all help is greatly appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems splicing Asterisk with a TE405P between Arcor E1 PRI and Ericsson Business Phone 250
Hi,
please make a pri debug span 2 of a call from the PBX to *
and show us the contents.
best regards
Klaus
--
Klaus-Peter Junghanns
Am Dienstag, den 25.01.2005, 22:39 +0100 schrieb Frank Sautter:
hi,
i'm having problems getting asterisk spliced between an E1 PRI (german
Telco Arcor) and an Ericsson Business Phone 250 digital PBX.
The Asterisk Server has a TE405P with it's port 1 connected to the E1
PRI provided by our telecommunications provider Arcor and port 2
connected to the E1 PRI of our Ericsson BP250.
the setup before:
Arcor TelCo PRI(E1) Ericsson BP250 PRI(E1)
the setup desired with asterisk spliced in:
Arcor TelCo PRI(E1) P1 asterisk P2--- Ericsson BP250 PRI(E1)
receiving and making calls between asterisk and the outside (arcor)
works so far (not entirely tested yet), but making calls from the
ericsson PBX to the asterisk server and routing them through to the
arcor PRI is not working.
the message i get when making a call from the ericsson pbx is:
Extension '' in context 'pri-ericsson' from '123498765' does not exist
obviously the ericsson pbx is not sending the dialled number on the pri
(but the calling number is set correctly)
as there is very limited time for me to play around with the parameters
in the asterisk config files (as the ericsson is in production use), i
hope the community can help me.
i think zaptel.conf is OK, as the LEDs are all green and the
communication between the all devices is working.
do i have to make changes on the ericsson PBX or in the zapata.conf?
regards
frank
here are some fragments of my config files:
/etc/zaptel.conf
# TE405P quad PRI(E1)
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47
span=3,2,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78
span=4,0,0,ccs,hdb3,crc4
bchan=94-108,110-124
dchan=109
loadzone=nl
defaultzone=nl
/etc/asterisk/zapata.conf
[channels]
language=de
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.8
txgain=0.8
callgroup=1
pickupgroup=1
immediate=no
context=pri-external
group = 1
signalling=pri_cpe
channel = 1-15,17-31
context=pri-ericsson
group = 2
signalling=pri_net
channel = 32-46,48-62
context=pri-loopin
group = 3
signalling=pri_cpe
channel = 63-77,79-93
context=pri-loopout
group = 4
signalling=pri_net
channel = 94-108,110-124
/etc/asterisk/extensions.conf (just the part of it that matters)
[pri-external] ; calls from the telco
include = durchwahl
exten = s,1,Answer()
exten = s,2,Dial(Zap/g2/${EXTEN})
exten = s,3,Hangup()
[pri-ericsson] ; calls from the ericsson BP250 to asterisk
include = durchwahl
exten = s,1,Answer()
exten = s,2,DigitTimeout,2
exten = s,3,ResponseTimeout,10
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Congestion
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Re: [Asterisk-Users] PRI - ISDN RESTART before connect
Hi, asterisk does not send the RESTART message, the switch sends the message right after the SETUP ACKNOWLEDGE message, note the s in the pri debug output. regards Klaus Am Freitag, den 21.01.2005, 11:09 +0100 schrieb hanson: When an outgoing call is dialed a SETUP isdn message is send to the telco's switch. The teleco's switch answers with an SETUP ACKNOWLEDGE message. In my special case * sends a RESTART message right after SETUP ACKNOWLEDGE , due to unknown reasons? This causes the channel to be restarted and the call establishing canceled ! ~-- Called r1/103330037x Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32932/0x80A4) (Terminator) Message type: SETUP ACKNOWLEDGE (13) [18 03 a9 83 88] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 8 ] - -- Processing IE 24 (cs0, Channel Identification) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 88] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 8 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI - ISDN RESTART before connect
Am Freitag, den 21.01.2005, 13:48 +0100 schrieb Hannes Kepplinger: Klaus, thanks for your quick reply. I thought that Originator or Terminator shows the direction. Danke, Hannes Hannes, yes it does, but you also have to look at the call reference. Basically the RESTART message from the switch is an outgoing call, that's why you find originator in there. best regards Klaus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot start asterisk - CAPI issues
Hi, Usually, you also need to load a firmware (with eiconctrl). Check out behind your card, when succesfully started, LEDs are turned on. There's no leds on this ISDN card... it's an old Eicon Diva 2.01 S/T that I got for 20 euros or so :-) This card does not have CAPI drivers. Only the Eicon Diva SERVER cards have capi drivers. Cheers, Jean-Michel. best regards Klaus-Peter Junghanns ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debian Sarge, ISDN, CAPI and Asterisk blues
Hi Joost, the W6692 based cards do NOT have capi drivers. At least not with isdn4linux, maybe it would work with the mISDN drivers. I have a W6692 card laying around on my desk (thanks voidptr :) ), a zaptel driver for that chipset is planned, but of course other things are more important. ;) best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Sa, 2004-09-25 um 10.12 schrieb Joost Kraaijeveld: Hi all, I am trying to get my Debian Sarge to work with 2 Winbond W6692 chipset based ISDN cards and Asterisk 1:0.9.1+1.0RC1-8. I have installed CAPI and chan_capi (all latest testing versions). If I start asterisk I get: chan_capi.c:2635 load_module: CAPI not installed. lsmod | grep capi gives: capi 17472 0 capifs60242 capi kernelcapi46496 1 capi Anyone any suggestions of where to look? Anyone a working asterisk with ISDN on Debian? Groeten, Joost Kraaijeveld Askesis B.V. Molukkenstraat 14 6524NB Nijmegen tel: 024-3888063 / 06-51855277 fax: 024-3608416 e-mail: [EMAIL PROTECTED] web: www.askesis.nl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Absolutely minimal Asterisk PSTN gateway
Am Sa, 2004-09-25 um 14.31 schrieb Arik Funke: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello together, I am setting up a communication server which should also act a very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500 MB usb memory stick. What is the ABSOLUTE minimum space requirements for ~ running asterisk to work as gateway between isdn and lan? 50MB or 1 GB?(I would compile, configure, etc. on a separate machine and then copy everything to the flash device.) Cheers, Arik Hi, 22 MB zipped for an *, postfix, router, traffic shaper, sshd. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] German Termination and DIDs
Hi, if i understand german telco regulations right (even for a german that's not an easy task...) then a provider may not assign a DID to a non-local client. This would mean that a provider in Berlin may not assign a DID to a client in Munich. So, assigning german DIDs to foreign clients would not be legal at all. Yeeehahh, regulations rule! :-) best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Sa, 2004-09-25 um 22.32 schrieb Eric Jacksch: Does anyone know of a company that provides German DIDs (preferably Berlin) and termination of calls to Germany at reasonable rates? Thanks, Eric [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pickup any call
Am Mi, 2004-08-18 um 15.36 schrieb Andrew Kohlsmith: On Wednesday 18 August 2004 08:34, Altus Snyman wrote: sorry,using the vpb.conf so card like voicetronix openline 4 card. Sorry my bad That I'm not sure -- I have never used a Voicetronix card. The callgroup/pickupgroup stuff is in zapata.conf though, but I believe similar mechanisms exist in iax.conf and sip.conf. You might want to take a look at app_pickup which is part of bristuff. It is a channel independent call pickup and call stealing application which is used in the normal dialplan, like: exten = *8,1,PickUP() ; uses the pickupgroup of the calling channel or exten = *8,1,PickUp(1) ; pick up from group 1 best regards Klaus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRI and E1 in same system
Hi Scott, make sure that ztcfg is only run once. Modprobing the e100p driver probably triggers this automatically. I am still investigating why qozap/zaptel becomes unhappy when ztcfg is run twice. If you want me to take a look via ssh, let me know. best regards Klaus Am Do, den 12.08.2004 schrieb Scott Stingel um 12:40: Hi- Anyone using the Junghann's quad BRI card and the Digium E100P in the same system? I'm having a configuration problem where I can configure the cards one at a time (with the appropriate drivers loaded) in a system, but when I try them both together, neither will work. They both work fine one at a time. Probably has something to do with the channel numbering. I've tried numbering the channels with the E1 first (which produces lots of modprobe errors), and then with the BRI span's first, which produces no modprobe errors, but doesn't work. Here is the latter configuration: ZAPTEL.CONF (excerpt): # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami # E1 definition: span=5,0,0,ccs,hdb3,crc4 #BRI's: bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 #E1: bchan=13-27,29-43 dchan=28 --- ZAPATA.CONF (excerpt) switchtype = euroisdn ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp ; define 4 BRI's: pridialplan = unknown prilocaldialplan = unknown echocancel = yes context=incoming group = 1 ; S/T port 1 channel = 1-2 group = 2 ; S/T port 2 channel = 4-5 group = 3 ; S/T port 3 channel = 7-8 group = 4 ; S/T port 4 channel = 10-11 ; E1 - for output to external Dialogic only ; we are pri_net in this case group = 9 pridialplan = unknown signalling=pri_net channel = 13-27,29-43 -- STARTUP MODPROBES, ETC: #following for Quad BRI system: cd /usr/src/bri/bri-stuff.0.1.0-RC2g/qozap modprobe -v zaptel /var/log/asterisk/modprobe.log sleep 1 insmod -v qozap.o /var/log/asterisk/modprobe.log sleep 1 # following for single E1 system modprobe -v wct1xxp /var/log/asterisk/modprobe.log sleep 2 ztcfg -vv /var/log/asterisk/modprobe.log sleep 3 echo /var/log/asterisk/modprobe.log --- Thanks for any help! Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Puzzled by CapiCD (call deflection to mobile phone)
Hi, Am Di, 2004-07-27 um 13.30 schrieb Loek Gijben: This says Call Deflection can only redirect an incoming call to another ISDN number. Ah well, this is not entirely true: ;) So what is the trick to deflect to a mobile phone, especially when both B channels are busy? You need to have a separate ISDN line for this, or do you set up a spare MSN number to redirect to a cellphone, need a VoIP-GSM gateway, or what? exten = s,1,capiCD(YOURmobilePhoneNUMBER) or have a deflect= line in capi.conf (make sure you enable the call deflection on busy support in the Makefil!). best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bri-stuff NT mode
Hi, modes= is a bitmap, 1 means NT mode, 0 means TE mode. So if you have 2 cards and the second should be NT then use: insmod zaphfc.o modes=2 if both are in NT mode then use: insmod zaphfc.o modes=3. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-07-26 um 10.16 schrieb Petr Grussmann: If I use 2 card and need NT mode on both 1card is in NT mode and second in TE mode after change zaphfc.c is change both to NT mode but not recognized parameter modes=1 or modes=0 sorry but I not know good C languages for better change -) need parameters for change if I have more card in Linux box Best regards, Petr Grussmann Opavanet a.s. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Display and UUS IEs on PRI - Q.931 question
Hi, you can take a look at how bristuff does this (it only has to be enabled in chan_zap to actually forward the display IE, uncomment line 8007). Latest version of bristuff is 0.1.0-RC2g which works with todays CVS versions. You can find it at www.junghanns.net/asterisk/ best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ At 7:00 PM +0200 on 7/26/04, Martin Blatter wrote: During an outgoing call on the PRI (E1, Euro ISDN) my provider regularly sends INFORMATION type messages containing IE 40 (Display) elements with the current cost of the call in progress (see below for example). The same Display IE is also transmitted during a RELEASE. I would like to pass this (very useful, IMHO) information to the calling device (i.e. a SIP phone) using the SendText() command. How can I get this information out of libpri and into Asterisk? Also, during setup of a call, we sometimes get IE 45 (User-to-User) containing the name of the caller. This information could be used as the Name part of the caller ID. How can I get and set this information within Asterisk? Thanks for any pointers. regards martin Message type: INFORMATION (123) [ [28 [28 08 [28 08 46 [28 08 46 52 [28 08 46 52 2e [28 08 46 52 2e 20 [28 08 46 52 2e 20 30 [28 08 46 52 2e 20 30 2e [28 08 46 52 2e 20 30 2e 32 [28 08 46 52 2e 20 30 2e 32 30 [28 08 46 52 2e 20 30 2e 32 30] Display (len= 8) [ FR. 0.20 ] -- Processing IE 40 (cs0, Display) -- Martin A. Blatter | lic. oec. publ. Wirtschaftsinformatiker | IT-Leiter OLMeRO AG | Europastrasse 30 | CH-8152 Glattbrugg | Switzerland [EMAIL PROTECTED] | phone +41 44 200 44 50 This is not an exact answer to your question, but might point you in the right direction if you wanted to make your own patch for incorporating UUI q.931 information into your system. If you make a patch, please submit it to http://bugs.digium.com/ since there are others who would find this useful. http://lists.digium.com/pipermail/asterisk-dev/2003-September/001751.html JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN, AVM C4, HFC-cards and echo
the hfc-pci cards use the same echo cancelation (in software) that any zaptel device uses. Am Do, 2004-07-08 um 09.47 schrieb Peer Oliver Schmidt: [interfaces] msn=123456 echosquelsh=1 make that echosquelch=1 incomingmsn=* controller=1 softdtmf=0 context=default ;echocancel=yes ;echotail=64 ;deflect=12345678 devices=2 callgroup=1 best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point DDI
I'm taking a slight tangent here, but stay with me. It looks like there are three methods of using HFC-S based ISDN BRI cards with *. Capi (via capi.conf), zaphfc (via zapata.conf), and isdn4linux (via modem.conf). Why, and which one is better and for which reasons? zaphfc is the best solution, because you get a zaptel timing source, NT mode support and echo cancelation! capi would be the next option, but then you would need to use mISDN which is not as stable. isdn4linux shouldnt be used at all. It is good for data application but unusable for voice (you will enjoy lots of echo and an increasing latency). -- best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point DDI
Am Mi, 2004-06-30 um 12.29 schrieb Holger Schurig: ASUS HN 100 ST D 128K (i think it has winbond chip) If you think then probably no one can say something for sure. Use lspci and post the output for this card. Just read what it says on the chipset. If it is a Winbond 6692 then just wait a little. A zaptel driver for that one is already in the works -- best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point DDI
Am Mi, 2004-06-30 um 17.21 schrieb Tomaz: ok, here is something to add .. correct me if I'm wrong! ok, i will! ;-) chan_capi: more features (early dial, call deflection, ISDN hold retrieve etc), stable, comes with echosquelch, works only with cards that have CAPI driver support; far more popular among Asterisk users compared to isdn4linux; can't detect if another application outside Asterisk is already using a specific ISDN channel (?) -DDI - p2p point2 point is NOT working It does not work with the binary only AVM Fritz card driver. A pure software limitation to sell their active cards All other capi cards support point-to-point (even the AVM Fritz card with mISDN's capi layer). -- best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip to isdn-capi call problem
Hi Tomaz, make sure you disable the G723.1 codec in your SIP device, asterisk does not support G723.1. Use G711 (alaw, ulaw)! best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-06-28 um 10.52 schrieb Tomaz: anyone has idea what problem can be here, something with codec but i have today CVS version and grandstream phone with 1.5.0 firmware.I try to change codec in phone and also in asterisk-sip.conf but the same. What can be problem ? tnx, Tomaz *CLI -- Executing Dial(SIP/102-767c, CAPI/2:5) in new stack -- Called 2:5 -- CAPI[contr1/2003002]/0 is making progress passing it to SIP/102-767c Jun 28 10:51:21 NOTICE[278545]: channel.c:1654 ast_set_read_format: Unable to find a path from G723 to ALAW Jun 28 10:51:21 NOTICE[278545]: channel.c:1621 ast_set_write_format: Unable to find a path from ULAW to G723 -- CAPI[contr1/2003002]/0 is ringing Jun 28 10:51:21 WARNING[278545]: chan_sip.c:1788 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 8/4) Jun 28 10:51:21 WARNING[278545]: channel.c:1485 ast_prod: Prodding channel 'SIP/102-767c' failed Jun 28 10:51:21 NOTICE[278545]: channel.c:1621 ast_set_write_format: Unable to find a path from SLINR to G723 Jun 28 10:51:21 WARNING[278545]: indications.c:76 playtones_alloc: Unable to set 'SIP/102-767c' to signed linear format (write) -- CAPI Hangingup == Spawn extension (from-sip, 9, 1) exited non-zero on 'SIP/102-767c' http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Protocol Error (6) using Zaphfc
Hei,
please never try to dial out on a particular b channel, you have to dial
out on a zaptel group which includes both b channels of the BRI line.
In a p2mp setup YOU cannot know which b channel will be chosen!
exten = _X.,1,Dial(ZAP/g1/${EXTEN})
will do(note the 'g')
best regards
Klaus
Am Mo, 2004-06-28 um 12.45 schrieb nrb:
Hi!
Has anybody seen anything like this using zaphfc?
On outgoing calls (via isdn) , the line gets hung-up as soon as the
called
party answers.
As seen below i get some protocol error (6) - but i'm not sure if this
is
related to the hang-up which apparently comes a little earlier?!
Incomming calls on the isdn (zaphfc) interface is working just fine
(P.S. what about the D-channel going up down all the time - is that
normal? )
Kind Regards
NRB
Setup
Bri-stuff - 0.0.20
Asterisk CVS-HEAD-06/23/04-15:45:48 built by
[EMAIL PROTECTED] on a
i686 running Linux
Zapata.conf:
[channels]
switchtype = euroisdn
; p2mp TE mode
signalling = bri_cpe_ptmp
; p2p TE mode
;signalling = bri_cpe
; p2mp NT mode
;signalling = bri_net_ptmp
; p2p NT mode
;signalling = bri_net
pridialplan=local
prilocaldialplan=local
echocancel=yes
immediate=yes
group = 1
context=demo
channel = 1-2
Zaptel.conf:
loadzone=nl
defaultzone=nl
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
Example where a sip client (2203) is calling 7024
From Asterisk:
== D-Channel on span 1 down
== D-Channel on span 1 up
-- Executing Dial(SIP/2203-5779, Zap/1/7024) in new stack
-- Making new call for cr 135
Protocol Discriminator: Q.931 (8) len=32
Call Ref: len= 1 (reference 7/0x7) (Originator)
Message type: SETUP (5)
Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer
capability:
Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
Ext: 1 User information layer 1: A-Law (35)
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0,
Exclusive
Dchan: 0
ChanSel: B1 channel
]
Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user number passed network
screening
(1) '2203' ]
Called Number (len=11) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '7024' ]
Sending Complete (len= 0)
-- Called 1/7024
Protocol Discriminator: Q.931 (8) len=7
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: CALL PROCEEDING (2)
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0,
Exclusive
Dchan: 0
ChanSel: B1 channel
]
-- Processing IE 24 (Channel Identification)
Protocol Discriminator: Q.931 (8) len=12
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: ALERTING (1)
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0
Location: Network beyond the interworking point (10)
Ext: 1 Progress Description: Inband information or appropriate
pattern now
available. (8) ]
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0
Location: Network beyond the interworking point (10)
Ext: 1 Progress Description: Unknown (1) ]
-- Processing IE 30 (Progress Indicator)
-- Processing IE 30 (Progress Indicator)
-- Zap/1-1 is ringing
Protocol Discriminator: Q.931 (8) len=15
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: CONNECT (7)
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0
Location: Public network serving the remote user (4)
Ext: 1 Progress Description: Unknown (4) ]
Time Date (len= 5) [ 04-06-28 11:58 ]
-- Processing IE 30 (Progress Indicator)
-- Processing IE 41 (Date/Time)
Protocol Discriminator: Q.931 (8) len=4
Call Ref: len= 1 (reference 7/0x7) (Originator)
Message type: CONNECT ACKNOWLEDGE (15)
-- Zap/1-1 answered SIP/2203-5779
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate
Connect
Request
Protocol Discriminator: Q.931 (8) len=8
Call Ref: len= 1 (reference 7/0x7) (Originator)
Message type: DISCONNECT (69)
Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location:
Private network serving the local user (1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]
-- Hungup 'Zap/1-1'
== Spawn extension (intern, 7024, 1) exited non-zero on
'SIP/2203-5779'
Protocol Discriminator: Q.931 (8) len=4
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: RELEASE (77)
-- Channel 1, span 1 got hangup
Protocol Discriminator: Q.931 (8) len=11
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: RELEASE (77)
Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location:
Public network serving the local user (2)
Ext: 1 Cause: Recover on timer expiry (102), class = Protocol Error
(6) ]
Cause data 0: 38 (56)
Cause data 1: bb (187)
Cause data 2: 5e (94)
-- Processing IE 8 (Cause)
NEW_HANGUP DEBUG: Calling q931_hangup,
Re: [Asterisk-Users] TE410P / Eicon PRI
We would think about having 2 servers : Server A : Asterisk PRI card (Digium TE410P) Server B : Fax server PRI card (Eicon PRI30M) Call --- TE410P/1 --- Asterisk Extension --- Voice ? --- Voicemail or Dial Fax ?--- TE410P/2 crossover to --- Server B (Eicon PRI) save 10k EUR and use spandDSP (www.opencall.org) for fax instead of the second server with the Eicon PRI card. Michael best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P / Eicon PRI
i'dd like to but is it stable enough for production (receiving over 500 faxes a day ?) i think it is. at least i know someone who is using it in production on a Digium E1 card. If everything else fails you can buy that eicon card later on in the worst case. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P / Eicon PRI
better send the EUR 10k (not $10k... :) ) to the author of spandDSP. Nobody needs HylaFAX for receiving faxes. Converting a tiff to pdf and storing it somewhere is not rocket science. ;) best regards Klaus Am Fr, 2004-06-18 um 17.08 schrieb Lee Howard: If you would rather use HylaFAX instead of spandsp and have $10K to throw around, then may I suggest hiring an Asterisk channel author to write a T.38-supporting channel driver? That way you could just use t38modem with HylaFAX, and you wouldn't need all the duplicate hardware. Lee. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P / Eicon PRI
Am Fr, 2004-06-18 um 17.53 schrieb Darren Nickerson: You don't even need spandsp - fax is dead, remember? ;-) Why do YOU sell hylafax servers then? ;) best regards Klaus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P / Eicon PRI
Am Fr, 2004-06-18 um 19.56 schrieb Lee Howard: Firstly, I'm not just talking about receiving faxes. If my choices are between HylaFAX and spandsp and if I want outbound queueing and a client-server interface for networked usage, then spandsp will not cut it alone. So yes, anyone who wants these features will need to use HylaFAX. And to use HylaFAX with Asterisk currently one must send the fax calls to an FXS port and then to a HylaFAX-controlled modem. Theoretically chan_capi could also be modified for fax support, since that is already part of the CAPI specs. But spanDSP works for all channel types so i dont see the need for this. For outbound spooling pbx_spool is your friend. If you want to take total control of the spooling yourself you can also build something very nice and scalable with the manager interface. This is not a pretty configuration, I completely agree. And, I completely agree that there are a myriad of beautiful ways to do this, in theory. But the coding does not exist for those to be reality. So unless someone wants to code it or pay to have it coded, then those who want outbound queueing and a client-server interface must put up with the cumbersome configuration. I agree that the hylafax clients are really nice and very useful. Furthermore, even if you assumed that spandsp was as stable as HylaFAX, there is a vast feature-set difference between them as far as the faxing itself goes. Steve has already made it clear that he sees no future in fax, and that he does not intend to bridge that feature-set gap at all. Correct me if I am wrong, but hylafax and spanDSP are two totally different pairs of shoes. Hylafax relies on the modem device to actually provide the fax capability. SpanDSP is pure software solution. You can fax with any Asterisk channel driver even VoIP. Apart from the missing network client you can build any feature you can dream about with Asterisk. Oh, and btw, i receive all my faxes with capi4hylafax and HylaFAX of course, just because SuSE comes with such a nice configruation tool for it (and i am lazy!). :) best regards Klaus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] embedded Asterisk
Hi, Actually, you the Geode CPU mentioned below is a 5x86 (486 platform) at 233 MHz. If you take Pebble (http://www.nycwireless.net/pebble/), which is a downstripped Debian ( 64 MB) on a readonly ext2 filesystem, you should be grand. Installing asterisk + some extra stuff will probably require, that you have at least a 128MB or 256MB flash or so. Dont go for stripped down but complete distributions which include a lot of stuff that you dont need, e.g. gcc. Go for a rescue system, like i used the SuSE rescue system (14 mb), then you can add what you need (sshd,...) and compile asterisk on another box and then just copy it. My compressed ramdisk image is 32 mb, including all voice prompts and some mp3s for MOH. There are actually quite some board around on that CPU, like Soekris, pcengines and i think also Mikrotik at prices from 120EUR and up. I just put together the demo system for Linuxtag: - Via EPIA 5000 (C3-533), EUR 80,- - Morex case with external power supply, EUR 80,- - some old 256 mb SDRAMM - 128 MB USB memory stick, EUR 30,- - 1 quadBRI (could also easily handle an octoBRI, or a PRI card, with the dual riser pci card you can use 2 cards) The C3-533 is an i586 CPU. According to show translation it needs 30 ms for transcoding 1 channel from g711 to gsm (and vice versa). So, neglecting any overhead caused by channel handling it could transcode 30 channels to gsm. Linux BIOS has support for the EPIA boards, so you can speed up booting very much and also disable the VGA port (very useful for production deployments). I'm running pebble on a pcengines board, just needed to customize the kernel a bit, haven't been testing asterisk on that yet, but i definatly will in the sooner future. Kind regards, Martin List-Petersen martin (at) list (dash) petersen (dot) net best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMoE Question
TDMoIP is nothing else like IAX2 with trunking, i would say. And a compression of 16/1 (payload bandwidth!) sounds like g723.1 to me. Just a Question. I would like to know if TDMoE follows specifiaciones of TDMoIP RAD protocol that says that there is a compression of 16/1 when you do TDMoIP. Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_Capi 0.3.4
please update to 0.3.4a. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-06-14 um 16.42 schrieb Jason Williams: Just tried compiling chan_capi 0.3.4 under CVS Head and get the following errors. chan_capi.c:60: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:61: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRI In the states
Daniel, no you are not stupid. It's just that very few people have had a BRI experience in the US. The only CAPI card with support for NI-1 is the Eicon DIVA Server (single BRI or four BRI). They work with NI-1 BRIs and chan_capi. And chan_capi supports the active echo cancelation on the Eicon cards. If you can wait for a while then you will be able to use the very cheap hfc-pci based ISDN cards in the US, too (or our quad and octoBRI cars). NI-1 support is already on my to-do list for the zaptel BRI support, but i cannot give you an ETA yet. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-06-07 um 16.22 schrieb Daniel Jimenez: Daniel Jimenez wrote: Hi all. I've ordered a TDM400P with 4 FXO, but after using my X100P I'm thinking about returning the TDM400P because of bad echo issues. If I do get the echo issues I'll look at digital options. My question: Is anyone using ISDN (BRI) in the states? I've heard ISDN4LINUX devices suffer bad echo but chan_capi works great. All the chan_capi cards I find though are for overseas (ie europe etc). Would I be better off looking at a fractional PRI? I'm only using 4 lines right now. I think a fractional PRI would be over kill. No one has any comments on this? No recommendations, or you are stupid for trying that or anything? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi and DDI (Anlagenanschluss)
Hi, you can use the AVM Fritz card in P2P mode, if you use the new mISDN capi layer with chan_capi. You dont have to rely on a kernel-tainting module anymore. Of course it makes sense to use a hfc-pci card instead, since it will provide your box with zaptel timing and echo cancelation! best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-06-07 um 16.32 schrieb Julian Pawlowski: The modules are from Linux 2.4.26, fcpci is from 03.11.02. I remember that it's not possible to have an AVM Fritz card on an PTP mode ISDN line. I think cards with HFC chipset are able to do so. Of cause you could also use an active card with CAPI driver ;-) Regards, Julian Pawlowski Verschicken Sie romantische, coole und witzige Bilder per SMS! Jetzt neu bei WEB.DE FreeMail: http://freemail.web.de/?mc=021193 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi and DDI (Anlagenanschluss)
Am Mo, 2004-06-07 um 17.33 schrieb Holger Schurig: Of course it makes sense to use a hfc-pci card instead, since it will provide your box with zaptel timing and echo cancelation! Still waiting on this card. Hehe, eventually I'll even get your quad-card. But beforehand I have to demonstrate to management that it VOIP-PBX actually works :-) if you need some help with demonstration we can set up an iax2 account together with a Berlin DID for testing. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan Capi Audio Quality Issue...
Am Di, 2004-06-01 um 14.42 schrieb Stefano Finetti: Well, i think i've solved the problem by myself :-) I had to change a line in chan_capi_pvt.h: /* was : 130 bytes Alaw = 16.25 ms audio not suitable for VoIP */ /* now : 160 bytes Alaw = 20 ms audio */ /* you can tune this to your need. higher value == more latency */ #define AST_CAPI_MAX_B3_BLOCK_SIZE 160 Putting the AST_CAPI_MAX_B3_BLOCK_SIZE to 130 (16.25ms audio) solved the problem. I forgot to mention that i'm using Snom105 phones. It seems that with GS BT101 with Ilbc firmware the value 160 works fine, but with snom it introduces an ugly distortion and choppy audio. This is really surprising. What codec are you using on the Snom? Recompiled using 130 as value, and the sound now is just really fine. A little question to Kapejod if he reads this... Is it possibile to put an even littler value here? I've tried to use 80 (= 10ms audio) but it makes impossible to start * cause it can't load chan_capi.so module. the minimum size for a DATA_B3_BLOCK in capi 2.0 is 128 bytes. Regards, -- best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference Server
Hi, take a look at zaprtc (which generates the zaptel timing out of your pc's realtime clock) or ztdummy (which uses an usb-uhci controller to generate the timing). best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Do, 2004-05-27 um 17.58 schrieb pesb: Hi there, I need to implement a SIP Conference Server. I've saw that asterisk has an application called meetme. But, it says that A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY. Is there any other way to implement a conference server without the need of having a ZAPTEL Interface? I need my conference server to work only with my SIP Phones. thanks in advance, Pablo Salinas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] outgoing MSN on zaphfc
Hi,
like any other zaptel device, zaphfc uses the callerid from the
originating channel. If you want to override that callerid use:
exten = _X.,1,SetCallerID(MyMSN)
If you want to restrict the outgoing callerid (CLIR) make sure
you have usecallingpres=yes in zapata.conf and use:
exten = _X.,1,CallingPres(32)
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/
Am Mi, 2004-05-26 um 20.04 schrieb Thomas Niesel:
On Wed, May 26, 2004 at 05:54:52PM +0200, Julian Pawlowski wrote:
Hi Thomas!
you have to set the MSN this way for zaphfc when you use the dial command:
exten = _0Z.,5,Dial(CAPI/MyMSN:${EXTEN},90,mT)
zap=capi???
For Capi its clear but...
Maybe my mistake but I have zaphfc in TE Mode connected to TelCo and like
to use different MSN via the dialplan.
It could be that there is a code like *55(MY_MSN)# to prefix the call.
But I'am not shure if euroISDN has such thing.
Perhaps I'am totaly wrong??
It works with i4l, capi why not having the choice with zaphfc? :)
Of course you have to set MyMSN to your MSN.
Regards,
Julian
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Re: [Asterisk-Users] quadBRI and CallerID
Hi Pedro, please do a bri debug span X on the span with the BRI line. Look at the SETUP message, the incoming caller ID is in the Calling Party IE. If you do not see the caller ID in that IE then your telco is not providing incoming caller ID on your line, some telcos like to charge extra for such basic isdn features. -- best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mi, 2004-05-19 um 15.26 schrieb Pedro Vela: Hi, I have a Junghanns.net quadBRI PCI Card with Telefonica ISDN BRI line, and we have in zapata.conf usecallerid=yes and hidecallerid=no, but we have not the caller ID. Can I make some configuration to solve this? Thanks, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr-csv / Zaptel BRI / full number not stored (missing leading zeroes)
hi, do you have nationalprefix=0 internationalprefix=00 in your zapata.conf? best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am So, 2004-05-16 um 16.21 schrieb Frederic Olivie: Hi, I'm using a ZaptelBRI card. It works fine. But I have a small problem with call logs. The leading zeroes of the external calling party are not stored (e.g. : 0140302010 will be stored as 140302010). Same for international numbers for which 00 will be stripped out. I would not mind if the cdr record would give me an indication of the call's origin (national or international), but it does not. The goal here is to implement a basic missed call web service that would allow my users to generate a call back. -- Frdric Olivi (Alf) @ Club-Internet Don't SCREAM, It hurts my eyes ! Ne CRIEZ pas, a fait mal aux yeux ! Alf, March 2001 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digital Line Distortion
Hi Adam, what is your echocancel setting in zapata.conf for the PRI spans? I once noticed this distorted sound by using echocancel=256 (using mec2.h for echo cancelation). How about echocancelwhenbridged and echotraining? best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-05-03 um 13.43 schrieb Adam Goryachev: Damn, I forgot to describe the actual problem. Basically as someone I spoke to today described it, it sounds like you have one of those new digital pbx systems... In more detail, when he spoke, he heard his voice come back, but distorted. The louder the sound he made, the louder he heard himself (distorted). At all times, if I am on the tdm40b side, I hear 100% perfect audio quality in both directions. (Which is bad, because now the customer gets the bad sound, before it was just the staff...) Regards, Adam On Mon, 2004-05-03 at 21:28, Adam Goryachev wrote: Firstly, the problem... Ever since I installed and setup asterisk, I have had various problems, initially it was echo caused by the ISDN (isdn4linux) card I was using. Now I'll do the thing most people forget about... [SNIP] the rest of the quoted text! Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with bri-stuff.0.0.2rc20a
Hola, if you have overlapdial=no in zapata.conf then * will jump into the s extension on a NT span (this way you can use DigitTimeOut and ResponseTimeOut to make patterns like _X. work as expected.). So, either you create an s extension, e.g.: exten = s,1,DigitTimeOut(3) or you set overlapdial=yes in zapata.conf. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am So, 2004-05-02 um 23.36 schrieb FastJack: hi everybody, just upgraded my bri-stuff driver to 0.0.2rc20a. now i have a strange problem :-( i have immediate = no but when i pickup the phone i get : *CLI == D-Channel on span 1 up -- Extension 's' in context 'default' from '6294094' does not exist. Rejecting call on channel 2, span 1 i have started asterisk with -vvc so there should be a debug message if immediate mode was on. maybe anyone (klaus-peter) can help. i'm using a hfc-card in nt-mode. i'm not 100% shure but i think that my phone is using uk-tones (ring ...) since the update but all language-settings are nl. looking forward to get some help ;) thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk dials wrong numbers ?!?
Thomas, never ever dial out on an indiviual B channel. create a group in zapata.conf for your PRI, like: group=1 channel = 1-15,17-31 and then use: exten = 1000,1,Dial(ZAP/g1/1234) best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am So, 2004-04-25 um 22.54 schrieb Thomas Schroeter: Hi, I've got an important question: I use an E100P directly connected to PSTN, but it does not *really* work as it should be: exten = 1000,1,Dial(Zap/1/1234) BUT: It does NOT dial 1234 but it says in debug mode: -- Called 1/72976451 Apr 26 00:53:00 WARNING[10251]: chan_zap.c:5979 zt_pri_error: PRI: !! Facility message shorter than 14 bytes -- Channel 1, span 1 got hangup Apr 26 00:53:00 WARNING[25617]: app_dial.c:347 wait_for_answer: Unable to forward voice Apr 26 00:53:00 WARNING[25617]: app_dial.c:347 wait_for_answer: Unable to forward voice -- Hungup 'Zap/1-1' == No one is available to answer at this time There is one exception: One specific number is dialled, and in the client I hear the real outside dialtone, but it is definitely not the number I wanted to dial! So it dials ONE number, but the wrong one...?!? Who can help...? Regards, Thomas --- Thomas Schroeter // +49-175-4624147 // +49-40-72976451 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel 536ep as a FXO?
Hi Am Mo, 2004-04-19 um 16.50 schrieb Jeremy Hall: I remember seeing somewhere that you can use a program (part of the zt suite if I remember correctly) to view the audio levels on the FXO card like an on-screen vu meter. I can use that and dial up my telco milliwatt test number and adjust accordingly. I asked where that tool was on the IRC channel, but they seemed to not know either. I have searched as I know I saw it, but can't find it again. That would be ztmonitor, i guess: silverbox:/usr/src/build/rc20/zaptel # ./ztmonitor 2 -v Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI: This number has been disconnected
Hi, it seems like you are using the 'r' option of app_dial. This will fake ring indication and will not pass any audio until the call is answered. What does your dial extension look like? best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am So, 2004-04-18 um 17.09 schrieb [EMAIL PROTECTED]: All, When calling an invalid number using, I expect to hear: dooh-deeh-daah We're sorry you have reached a number which has been disconnected ... And that is indeed what I hear when I dial out from [*] using analog FXO, or VoicePulse or NuPhone. When I dial that same number trough the T1 / PRI interface however, I continually hear ringing, and then the call gets hungup. Any ideas anyone? It kinda annoys our users, since they like to *know* when they dial an invalid number. TIA, WW Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callerid + Zaphfc
Hi, bristuff 0.0.2rc20 will add support for HOLD/RETRIEVE, SUSPEND/RESUME and isdn transfers in an experimental way. It also features a zaptel that works on 2.6 (and does not freeze), together with optimized qozap drivers. Load tests have shown that it is possible to have 6 quadBRI cards in a decent P4 system ( 2.8 Ghz). Expect RC20 in the next 2 or 3 days. -- best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-04-12 um 22.47 schrieb Martin List-Petersen: On Thu, 2004-04-08 at 08:54, Martin Schenkelberg wrote: Thank you problem solved. I tried to use the (R) Button on my phone to place call on HOLD but Asterisk says something of PRI Error : Dont know how to post-handle message of Tye HOLD (36) Is this feature not implemented in Bri-Stuff ? Thanks again Both HOLD, CONFERENCE and others are not implemented. This is actually not he BRI stuff, but libpri that handles it, because they are generic ISDN features. I assume it would take a bit more to get that implemented. You can allways use the * parking system, (press #, transfer to 700). Kind regards, Martin List-Petersen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callerid + Zaphfc
Hi, use prilocaldialplan=local in zapata.conf. -- best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mi, 2004-04-07 um 09.59 schrieb Martin Schenkelberg: Hi all, i have an ISDN Phone connected to an HFC-S based card, all works fine but is i call the Phone from a SIP User Agent or over PSTN Line the Phones Display shows the correct CallerID but with a leading 0 . I cant find this in the config files, how can is solve this? Dialing Out with the ISDN Phone transmitts the correct callerid. Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: There is no FORK! (WAS Re: [Asterisk-Users] BETA RADIUS support for Asterisk)
This is why disclaimers are important for those who contribute patches. If there isn't a disclaimer, Digium can not include it in the proprietary version of asterisk. If they can not include it in the proprietary version, they tend to not allow it in their version of the GPL releases so they don't have to maintain a real proprietary fork as well as the GPL version. THERE IS NO FORK! There is a total of ONE (1) Asterisk source tree. well, if i understand the GPL right there has to be a fork. If there is a non-GPL licensed * it cannot contain the mysql friends funtionality which is in chan_iax2 and chan_sip since the mysql client libs are GPL only. Please correct me if i am wrong. best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MFE for TEI=76
Hi Jens, MFE for TEI=76 means that a layer 2 p2p connection has been established between your * and your telco's switch. This is a good thing! :) It looks like your telco is pulling down layer 2 when the line is idle (probably also layer 1 for power saving). Do you see a lot of card X span Y state FZ (A_ST_RD_STA = 0x1Z) messages in dmesg? Those indicate state changes on layer 1. I am currently cleaning up some things, so these messages will only appear if you have bri intense debug span Y enabled. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Sa, 2004-03-06 um 23.39 schrieb Jens P. Hansen: I have a very basic setup using some Sip phones and a QuadBRI adaptor. Everything seem to be running fine, however * spews out MFE for TEI=76 every 10 sec. or so, on the console. Why do I get this message and what may be done to elliminate it ? Kind Rgds ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tiny install with Solid State Storage
Hi, I am running * on a modified SuSE 9.0 rescue system. Total system including sshd, *, MOH and * prompts is 32 MB zipped. It expands to 52 MB on a 64 MB RAM disk. I boot it from a compact flash disk. The system is a 600 mhz transmeta crusoe with only 110mb ram. It is powerful enough to drive a quadBRI and so some GSM encoding/decoding. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-03-01 um 17.37 schrieb Matt: Hello John, I saw the wiki page on trustix, it said 296 megabytes, still a little big. I'm downloading trustix now to check it out though. Thanks -Matt - Original Message - From: John Bittner [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, March 01, 2004 10:32 AM Subject: RE: [Asterisk-Users] Tiny install with Solid State Storage I have a unit running Redhat 9 on a 1 gig flash card. Since a 1 gig flash card is expensive I am working on a unit running http://www.trustix.net/ on a 256meg flash card. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Monday, March 01, 2004 11:02 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Tiny install with Solid State Storage Hello All, I was wondering if anyone is successfully running asterisk on a system with solid state storage, such as a compact flash card? I'm looking for some pointers on doing this. Thanks -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tiny install with Solid State Storage
http://ftp.gwdg.de/pub/suse/i386/9.0/boot/rescue best regards Klaus Am Mo, 2004-03-01 um 18.38 schrieb Matt: Hello Klaus, Is it possible for me to download an image of the os or can you point me to the rescue disk that you used? - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, March 01, 2004 11:54 AM Subject: Re: [Asterisk-Users] Tiny install with Solid State Storage Hi, I am running * on a modified SuSE 9.0 rescue system. Total system including sshd, *, MOH and * prompts is 32 MB zipped. It expands to 52 MB on a 64 MB RAM disk. I boot it from a compact flash disk. The system is a 600 mhz transmeta crusoe with only 110mb ram. It is powerful enough to drive a quadBRI and so some GSM encoding/decoding. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-03-01 um 17.37 schrieb Matt: Hello John, I saw the wiki page on trustix, it said 296 megabytes, still a little big. I'm downloading trustix now to check it out though. Thanks -Matt - Original Message - From: John Bittner [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, March 01, 2004 10:32 AM Subject: RE: [Asterisk-Users] Tiny install with Solid State Storage I have a unit running Redhat 9 on a 1 gig flash card. Since a 1 gig flash card is expensive I am working on a unit running http://www.trustix.net/ on a 256meg flash card. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Monday, March 01, 2004 11:02 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Tiny install with Solid State Storage Hello All, I was wondering if anyone is successfully running asterisk on a system with solid state storage, such as a compact flash card? I'm looking for some pointers on doing this. Thanks -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Changes in capi.conf
Hi, You dont have to reboot your machine, you only have to restart asterisk, restart when convenient is a safe way to do this. Same thing as with the zaptel channel driver. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Di, 2004-02-24 um 13.51 schrieb Jan Larsen: I have notised that when ever I nake a change in capi.conf (from junghanns) I have to reboot the maschine before the thanges is activated. A reload does not do the thing. Is this behavior right ?? Regards. Jan Larsen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode
Hi, to clear things up again, the problem is a wrong syntax for the Dial appplication exten = 74341423,1,Dial(Zap/g2/74341423,r) This will use r as the timeout value, so it will hang up immediately, actually too quick for the isdn phone to bring up a p2p layer 2 connection (it is on my todo list to handle this too). exten = 74341423,1,Dial(Zap/g2/74341423,,r) This is what you want (not the second ,). best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Do, 2004-02-19 um 23.30 schrieb Armand A. Verstappen: Hi Ernst, On Thu, 2004-02-19 at 15:26, Ernst Lehmann wrote: use this: exten = 74341423,1,Dial(Zap/g2/74341423,r) snip Interesting fact is, that the ISDN-Phone on the NT line rings still, if the calling phone has dropped the call.. The same thing here. wkr, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode
Hi Ernst, use this: exten = 74341423,1,Dial(Zap/g2/74341423,r) instead of: exten = 74341423,1,Dial(Zap/5/74341423,r) -- best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Do, 2004-02-19 um 14.24 schrieb Ernst Lehmann: Hi, Does anyone operate Asterisk with zaphfc in NT-Mode successfully ?? I have the problem, that I could not contact my ISDN-Phone on such a channel. It rings, but If I pick up the phone, I only get a Hangup in the console Thanks for any clues on it... Here my setup: 3 HFC Cards. first card is TE, other two are NT mode (loaded zaphfc with modes=6) The ISDN-Phone is connected to the second-card. The ISDN-Bus is powerd by an old NTBA like described in the pbx4linux project Howto. /etc/zaptel.cfg span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,0,3,ccs,ami bchan=4-5 dchan=6 span=3,0,3,ccs,ami bchan=7-8 dchan=9 --- /etc/asterisk/zapata.conf - switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan=unknown echocancel=yes immediate=no group = 1 context=prod channel = 1-2 switchtype = euroisdn signalling = bri_net_ptmp pridialplan=unknown group = 2 context=prod channel = 4-5 - valid passage from extension.conf -- [prod] exten = 74341423,1,Dial(Zap/5/74341423,r) . --- The ISDN-Phone is configured to listen on the MSN 74341423 Here the log from the console: --- -- Executing Dial(Zap/2-1, Zap/5/74341423|r) in new stack -- Called 5/74341423 -- Hungup 'Zap/5-1' -- Accepting call from '8974341421' to '74341423' on channel 2, span 1 MFE for TEI = 64 -- Timeout on Zap/2-1 == CDR updated on Zap/2-1 -- Executing Goto(Zap/2-1, #|1) in new stack -- Goto (prod,#,1) -- Sent into invalid extension '#' in context 'prod' on Zap/2-1 -- Executing SetVar(Zap/2-1, starttime=1077196512) in new stack -- Executing Playtones(Zap/2-1, info) in new stack -- Executing Wait(Zap/2-1, 1) in new stack -- Executing Playback(Zap/2-1, invalid) in new stack -- Playing 'invalid' (language 'en') -- Executing Wait(Zap/2-1, 1) in new stack -- Executing GotoIf(Zap/2-1, 1?2:7) in new stack -- Goto (prod,i,2) -- Executing Playtones(Zap/2-1, info) in new stack -- Executing Wait(Zap/2-1, 1) in new stack -- Executing Playback(Zap/2-1, invalid) in new stack -- Playing 'invalid' (language 'en') -- Channel 2, span 1 got hangup == Spawn extension (prod, i, 4) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' - perhaps helpfull... some information from dmesg on loading of modules: - Zapata Telephony Interface Registered on major 196 zaphfc: start PCI: Found IRQ 5 for device 00:09.0 PCI: Sharing IRQ 5 with 00:04.3 zaphfc: card configured at mem 0xc8875000 fifo 0xc70e8000(0x70e8000) IRQ 5 HZ 10 0 zaphfc: ZTHFC1/0/1 zaphfc: ZTHFC1/0/2 zaphfc: ZTHFC1/0/3 zaphfc: registered zaptel device! zaphfc: Card 0 configured for TE mode zaphfc: resetting card. zaphfc: layer 1 state = F4 PCI: Found IRQ 12 for device 00:0a.0 zaphfc: card configured at mem 0xc8877000 fifo 0xc7158000(0x7158000) IRQ 12 HZ 1 00 zaphfc: ZTHFC2/0/1 zaphfc: ZTHFC2/0/2 zaphfc: ZTHFC2/0/3 zaphfc: registered zaptel device! zaphfc: Card 1 configured for NT mode zaphfc: resetting card. zaphfc: layer 1 state = F5 zaphfc: layer 1 state = F6 zaphfc: layer 1 state = G2 PCI: Found IRQ 10 for device 00:0b.0 PCI: Sharing IRQ 10 with 00:11.0 zaphfc: card configured at mem 0xc8879000 fifo 0xc7148000(0x7148000) IRQ 10 HZ 1 00 zaphfc: ZTHFC3/0/1 zaphfc: ZTHFC3/0/2 zaphfc: ZTHFC3/0/3 zaphfc: registered zaptel device! zaphfc: Card 2 configured for NT mode zaphfc: resetting card. zaphfc: bchan rx fifo not enough bytes to receive! (z1=518, z2=8191) zaphfc: bchan rx fifo not enough bytes to receive! (z1=518, z2=8191) zaphfc: 3 card(s) in this box. Registered tone zone 3 (Netherlands) zaphfc: layer 1 state = G3 --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi problem
Am Di, 2004-02-17 um 09.33 schrieb dfm: Hi to all I've mada up my mind and i tried to change from i4l to chan_capi, following some councelling from the gurus. I compiled it up, and when i try to load it in modules.conf, i get that wonderful message and Asterisk does not start: [chan_capi.so]Feb 17 09:21:40 WARNING[16384]: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group Feb 17 09:21:40 WARNING[16384]: loader.c:358 load_modules: Loading module chan_capi.so failed! Any idea? In modules.conf I have: noload = chan_modem.so load = chan_capi.so [global] chan_modem.so=no chan_capi.so=yes But in capi.conf i really don't know what exactly to put, i left it as it comes, but i don't know how to set this file up. Any one is a chan_capi guru Regards Diego here we go again... put: load = res_parking.so before load = chan_capi.so best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to interface to BRIs
Hi Jim, we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN. You can find more information about it at: http://www.junghanns.net/asterisk/page17.html best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-02-16 um 10.10 schrieb Jim Archer: Hi All... I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to interface to BRIs
Hi Jim, i forgot to mention that the drivers do not yet support NI-1, but will support it in the near future. Until then the only solution for you will be the Eicon Diva Server 4BRI-8M and chan_capi. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-02-16 um 10.53 schrieb Jim Archer: I forgot to mention, I am in North America. --On Monday, February 16, 2004 4:10 AM -0500 Jim Archer [EMAIL PROTECTED] wrote: Hi All... I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analogical FXO vs. BRI dialing speed
Am Mo, 2004-02-16 um 14.39 schrieb Jean-Marc V. Liotier: When dialing out, will a call be established significantly faster by an ISDN adapter such as an Eicon Diva server compared to an analogical FXO such as Digium's X100P ? Yes, ISDN uses digital signalling so call setup times on the last mile (from your NT1 to the telco switch) are close to 0. Also the callerID on incoming calls is available immediately with ISDN (with analog lines you usually get it after the first ring). best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Need to interface to BRIs
The FritzCard has CAPI drivers and does NOT provide zaptel timing. The quadBRI PCI has zaptel drivers and does provide zaptel timing. Am Mo, 2004-02-16 um 14.41 schrieb Master Abi: Does the Fritz!Card PCI and Quad BRI also provide timing like the Digium Zaptel cards? Matteo Brancaleoni wrote: Il lun, 2004-02-16 alle 12:49, Cees de Groot ha scritto: Klaus-Peter Junghanns [EMAIL PROTECTED] said: we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN. One thing I'd like to know about this card: Echo Cancellation? I've replaced by Fritz!Card PCI by a Diva Server 2M, and the difference is remarkable... since is zaptel based, it shares same zaptel routines for EC, as far as I know. Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax
Hi, make sure you have echo cancelation disabled on that zaptel channel. regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Hi All, My asterisk system is running well but I can't send or receive faxes. I have an analogue fax plugged into a TDM400 connected to my ISDN 2e via an Eicon Diva. I am using G711.U - do I stand a chance of faxing or should I be doing it differently? Simon -- Simon Faulkner - Dedicated Programmes 01538 303 900 - 07771 845 326 http://dpnet.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN update
oh yes... i added callgroup support for chan_capi. That's why you have to load res_parking.so before chan_capi.so. So in modules.conf you need. load = res_parking.so load = chan_capi.so [global] chan_capi.so=yes best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ I tried make, make install. /usr/bin/asterisk -vvvgc and what I get is: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group loader.c:358 load_modules: Loading module chan_capi.so failed! what's wrong? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk act like a normal sip phone?
hi, just signed up and it works like a charm. :-) They even support g711 :) and multiple channels :) make sure you have in sip.conf: register = :[EMAIL PROTECTED]/extension in your context you will get the too many hops if you try to register with their proxy (proxy.de.sipgate.net). best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ I set up an account with sipgate yesterday evening and tried to use the above mentioned register in sip.conf * to login to sipgate. No luck so far. They use SER and I get 483 too many hops replies back from them. Any help is greatly appreciated. -Walter -- Walter Doerr =*= [EMAIL PROTECTED] =*= FAX: +49 2421 962001 The poor folks who only have 100MBytes of RAM five years from now may not be able to buffer a 16MB packet, but that's their tough luck. (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI: Early-B3 working with AVM-B1?
Hi Karsten,
are you sure your MSN is correct? If not T-Com will replace it
with your main MSN and probably will ignore the CLIR setting.
best regards
kapejod
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel: 1-700-157-8753
http://www.Junghanns.NET/asterisk/
Am Do, 2004-01-22 um 09.00 schrieb Karsten Wemheuer:
Hi,
here is an update to my own post to this list.
Following an information from Philipp, I testet this with an passive AVM
card, but the same things happen. What am I doing wrong?
Is there something wrong with my extension.conf?
without Early B3:
exten = _0X.,1,Dial(CAPI/@22715291:${EXTEN:1}|30)
with Early B3:
exten = _0X.,1,Dial(CAPI/@22715291:b${EXTEN:1}|30)
Thanks,
Karsten
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Re: [Asterisk-Users] ISDN p2p AVM Fritz Card
Hi Stephan, the passive AVMs do NOT support P2P. best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mi, 2004-01-21 um 14.39 schrieb Stadlbauer Stephan: hello! I'd like to use my asterisk-box on a p2p-line from my alcatel pbx. isdn4linux and capi works fine on a s0-line from the pbx, but i need more then 8 msns, cause i'd like to implement a voice-mail system and an extension to our old pbx with voip-phones. had anyone success using isdn4linux on a t0 (p2p) line (of course with an inexpensive passive card like my avm fritz!, cause i'm still in the testing phase, and don't like to buy expensive hardware at this point) ? if so - any information would be helpfully. by the way i'm from austria - so using euroisdn! thanxs in advance, stephan stadlbauer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie ISDN question
Hi Thorsten, the E100P is a PRI ISDN Card (S2M in Germany). You cannot connect phones to that card. The quadBRI card has 4 BRI ports that can individually be configured for TE mode (to connect ISDN lines) or NT mode (to connect ISDN phones). Please find the details at: http://www.junghanns.net/asterisk/page17.html best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ hi everybody, sorry for posting such a stupid question ;) i've managed to run asterisk* with my AVM fritz2.0 card and a some VOIP-softphones (SIP, H323). the functions of asterisk* really satisfied me ;))) now i want to run asterisk* istead of our old PBX. but it would be great to connect some phones directly to my box. how does a E100P from digium work. can i connect it to my ISDN-line and my internal phones (ISDN)? it would look like this: [PHONE2] / [PC]-[E100P] - [PHONE1] \ [ISDN-LINE] thank you for your help!!! thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie ISDN question
The quadBRI card is EUR 600, excluding VAT. best regards kapejod Hello kapejod, The quadBRI card has 4 BRI ports that can individually be configured for TE mode (to connect ISDN lines) or NT mode (to connect ISDN phones). Please find the details at: http://www.junghanns.net/asterisk/page17.html when are you going to release some pricing on the card? It just says But me!, but does not show you how... :) rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie ISDN question
Thorsten, theoretically you can connect 8 phones per port, but only 2 can be used at the same time. We advise to use 2 per port and in some scenarios 3 might be an option. So you can connect 8 ISDN phones to the quadBRI card. The drivers are still released as experimental and have some bugs. We are planning to be stable in about 2 weeks. The cards are in stock, so delivery will be fast. We ship with worldwide with UPS. best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ hi klaus-peter, thank you for your replay. btw: i am using you chan_capi already ;)) it works great!!! how many internel phones could be connected to this card? how stable is the driver (can i use it for a production-system)? sorry for all that stupid questions - i know linux and ip and pc-hardware but telephone-technics are all new for me. how long would delivery of that card take? thanks (oder besser gesagt: VIELEN DANK ;) ) thorsten - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 11:54 AM Subject: Re: [Asterisk-Users] newbie ISDN question Hi Thorsten, the E100P is a PRI ISDN Card (S2M in Germany). You cannot connect phones to that card. The quadBRI card has 4 BRI ports that can individually be configured for TE mode (to connect ISDN lines) or NT mode (to connect ISDN phones). Please find the details at: http://www.junghanns.net/asterisk/page17.html best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ hi everybody, sorry for posting such a stupid question ;) i've managed to run asterisk* with my AVM fritz2.0 card and a some VOIP-softphones (SIP, H323). the functions of asterisk* really satisfied me ;))) now i want to run asterisk* istead of our old PBX. but it would be great to connect some phones directly to my box. how does a E100P from digium work. can i connect it to my ISDN-line and my internal phones (ISDN)? it would look like this: [PHONE2] / [PC]-[E100P] - [PHONE1] \ [ISDN-LINE] thank you for your help!!! thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: newbie ISDN question
Hi, yes, for the home user it's still too expensive. Although it's really cheap if you compare it to other 4 BRI cards on the market. Currently i am polishing the driver for the hfc-s pci a chipset, which i used in numerous el-cheapo ISDN cards (street price around 30 EUR). This will bring zaptel BRI (and even NT mode) to the home user. :) best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Hi, On Wed, 14 Jan 2004 at 12:15, Klaus-Peter Junghanns wrote: The quadBRI card is EUR 600, excluding VAT. this looks like a great piece of hardware, but I think it's too expensive for home users like me who wouldn't really need more than one or two BRI ports. So do you have any plans for a singleBRI or doubleBRI version of this card, or maybe even a variant that comes with a single port preinstalled and three more ports can be added as needed via daughterboards like on the TDM400P? cu Reinhard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: newbie ISDN question
Is there a list of cards that use this chipset somewhere on the 'net? I've googled for it, but most pages only talk about cards based on the HFC-S chipset without listing brand and model names. Acer ISDN-Surf, Billion Bipac ISDN, Trust PCI ISDN Modem, D-LINK DMI-128+ to name a few ;-) regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple phonenumbers on one E1 PRI with Digium TE410P ?
Hi Jan, yes you can: [zap-in] exten = _49xxx,1,Goto(contextA) exten = _49xxx,1,Goto(contextB) regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Hi, one short question: Is it possible for the zaptel driver to deal with multiple phone numbers on one single E1 PRI line? I could make my carrier route +49 xxx a-zzz and +49 xxx b-zzz and others down one single PRI trunk to our asterisk box terminating in a Digium TE410P. Does the driver handle this and can I put calls coming in all on the same physical interface put into different contexts based on the dialed prefix? Thanks and Regards, Jan Baumann ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FS/OS Telephony Summit 2004
Hello * world, i will be attending the FS/OS Telephony Summit 2004 in Geilenkirchen from the 16th til 20th january. Together with Christian Richter i will be speaking about * on monday. And we will give an * tutorial on tuesday. I will be presenting some ISDN stuff there, including the quadBRI cards. If you will be there too and want to meet, just let me know. :) Details on the summit can be found at: http://www.guug.de/veranstaltungen/telephony-summit-2004/ best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Free Software/Open Source-Telephony-Summit 2004
Hi, since there will be people from around the globe it will all be done in English. regards kapejod Hi Philipp- Just out of curiosity, are these types of workshops generally conducted in German, or in English? Cheers Scott London Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Tuesday, December 16, 2003 3:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Free Software/Open Source-Telephony-Summit 2004 Hi, I just came across this annoncement, which is particularly interesting as it is only 25 min away from my place... :-) Anyway, I guess the core of this is targeted at developers mainly. Cheers, Philipp Free Software/Open Source-Telephony-Summit 2004 http://www.guug.de/veranstaltungen/telephony-summit-2004/ http://www.heise.de/newsticker/data/avr-16.12.03-000/ We are happy to announce that the first summit on Free Software/Open Source-telephony solutions is going to take place from January 16th till 20th in Geilenkirchen, Germany. The event will be divided into three parts: a developer workshop from January 16th to January 18th a conference day (January 19th) a tutorial day (January 20th) There will be an exhibition during the conference and the tutorial day. The developer workshop is free of charge and only-open for active developers in Free Software/Open Source telephony projects. If you are interested in participating, please contact Martin Schulte telephony- [EMAIL PROTECTED] During the one-day conference, the participating projects will give an overview about their current status and their future goals. In the morning of the tutorial day, people interested in using a particular software will get an in depth-introduction to installation, configuration and usage by their developers. In the afternoon, there's a big How everything works together-tutorial. Online-Registration is opened! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E400 or TE410 (digium) vs PRI 30M (Eicon)
Hi, the Eicons work fine with chan_capi, also the hardware echo cancelation works fine. regards kapejod Am Mo, 2003-12-15 um 17.38 schrieb Steven Critchfield: On Mon, 2003-12-15 at 09:58, Daniel ANDRE wrote: Hello, I would like to have some comparison between E1 cards from Digium and those from Eicon for a VOIP - ISDN Gateway. How does they compare on the echo cancel point of view? Is the echocancellation code for E400 good enough for production environment? The code for the Digium E1 is the same as used on any other Digium interfaces. It is in software and shared across all Zaptel hardware. There shouldn't be an echo at the E1 interface. The echo will be on either an analog end or in speaker bleed over to mic. Something you will also need to consider, what software will you be using? Digium cards work with asterisk, and I doubt that the Eicon E1 cards do yet unless they use the capi driver. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
