Re: [Asterisk-Users] Q.SIG support in CVS
--On February 18, 2005 11:16:11 -0600 [EMAIL PROTECTED] wrote: On Fri, Feb 18, 2005 at 02:18:37PM +0100, Kurt Bauer wrote: I just read thru the changelog.txt of the current CVS version and what catched my eye was the following line: 'Adding Q.SIG switchtype option to chan_zap' . But there is no sample config in zapata.conf for Q.SIG and no 'feature-list'. Does this exist anywhere or has anyone already has experience with * and Q.SIG and wants to share ?? Yeah, I've got some experience with it (I'm the one working on it :-) ). Right now we can do send/receive of DivertingLegInformation2 messages, message waiting indication activate/deactivate, and receive of calling name information. Oh, and of coure all your basic PRI stuff, such as call setup and teardown. So all I have to do is change the line 'switchtype=euroisdn' in zapata.conf to 'switchtype=Q.SIG' ??? I still haven't got the point how to tell my * Box that it is now Q.SIG aware :-o Thanks, br, Kurt ___ 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] Q.SIG support in CVS
Hi, I just read thru the changelog.txt of the current CVS version and what catched my eye was the following line: 'Adding Q.SIG switchtype option to chan_zap' . But there is no sample config in zapata.conf for Q.SIG and no 'feature-list'. Does this exist anywhere or has anyone already has experience with * and Q.SIG and wants to share ?? Thanks a lot in advance, best regards, Kurt ___ 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] supposable timing problem with TE100P
--On Thursday, November 04, 2004 04:41:53 PM +0100 Peter Svensson [EMAIL PROTECTED] wrote: On Thu, 4 Nov 2004, Kurt Bauer wrote: Is your timing source set correctly? If you are connecting to the pstn the pstn connection should be the primary timing source. connection is to a Ericsson MD110 wich is set as network, * is set as CPE. Have you set the span as the timing source? (second number in the span line in zaptel.conf). It was 0, I changed it to 1, which seems to work better, but I have to check the settings of the other side with our telephony guys What does the missed interrupts counter in cat cat /proc/zaptel/1 say? cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) Is there no IRQ miss counter for the E100P card? ??? I don't know. Thanks, Kurt ___ 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] supposable timing problem with TE100P
Hi list, every now and then I get the following message in my * logs: chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 As this is only a notice and voice worked quite well, despite the messages, I didn't bother. _But_, I wanted to try spandsp/fax (latest version) lately and whenever I send a fax to * the pages get chopped, which according to Steve Underwood points at a timing problem, either in * or Hardware. I don't know if the messages about FCS errors have smth. to do with the fax problems, but hope that someone out there has a clue ;-)) BTW, I see a lot of the following messages too: !! Unknown IE 49 (cs5, Unknown Information Element) !! Unknown IE 50 (cs5, Unknown Information Element) If any further information is needed to narrow the problem down please let me know. Thanks a lot in advance, best regards, Kurt ___ 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] supposable timing problem with TE100P
--On Thursday, November 04, 2004 03:19:56 PM +0100 Peter Svensson [EMAIL PROTECTED] wrote: On Thu, 4 Nov 2004, Kurt Bauer wrote: Hi list, every now and then I get the following message in my * logs: chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 As this is only a notice and voice worked quite well, despite the messages, I didn't bother. _But_, I wanted to try spandsp/fax (latest version) lately and whenever I send a fax to * the pages get chopped, which according to Steve Underwood points at a timing problem, either in * or Hardware. Is your timing source set correctly? If you are connecting to the pstn the pstn connection should be the primary timing source. connection is to a Ericsson MD110 wich is set as network, * is set as CPE. What does the missed interrupts counter in cat cat /proc/zaptel/1 say? cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) 5 WCT1/0/5 Clear (In use) 6 WCT1/0/6 Clear (In use) 7 WCT1/0/7 Clear (In use) 8 WCT1/0/8 Clear (In use) 9 WCT1/0/9 Clear (In use) 10 WCT1/0/10 Clear (In use) 11 WCT1/0/11 Clear (In use) 12 WCT1/0/12 Clear (In use) 13 WCT1/0/13 Clear (In use) 14 WCT1/0/14 Clear (In use) 15 WCT1/0/15 Clear (In use) 16 WCT1/0/16 HDLCFCS (In use) 17 WCT1/0/17 Clear (In use) 18 WCT1/0/18 Clear (In use) 19 WCT1/0/19 Clear (In use) 20 WCT1/0/20 Clear (In use) 21 WCT1/0/21 Clear (In use) 22 WCT1/0/22 Clear (In use) 23 WCT1/0/23 Clear (In use) 24 WCT1/0/24 Clear (In use) 25 WCT1/0/25 Clear (In use) 26 WCT1/0/26 Clear (In use) 27 WCT1/0/27 Clear (In use) 28 WCT1/0/28 Clear (In use) 29 WCT1/0/29 Clear (In use) 30 WCT1/0/30 Clear (In use) 31 WCT1/0/31 Clear (In use) br, Kurt ___ 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] SIP authentication problem
Hi, I have the following setup: E100P SER * - PBX This works just fine, except when there are users on both boxes (ie. SER and asterisk), whose usernames are the same, although the realm is different. An example: user '[EMAIL PROTECTED]' wants to call some extension in the PBX, but as user '[EMAIL PROTECTED]' exists too, * tries to authenticate the user, which it shouldn't do, at least I guess so. Shouldn't asterisk differentiate between the realms ie. [EMAIL PROTECTED] != [EMAIL PROTECTED] ? Find attached, the relevant part of the logged sip communication and the sip.conf. If you have any hints, please let me know. Thanks in advance, best regards, Kurt example sip.log Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Max-Forwards: 10 Record-Route: sip:[EMAIL PROTECTED];ftag=000cce3a7be800087fd8099f-62cc5396;lr=on Via: SIP/2.0/UDP 83.136.32.160;branch=z9hG4bK1cba.998c27f1.0 Via: SIP/2.0/UDP 131.130.220.101:5060;branch=z9hG4bK03521c86 From: Kurt Bauer sip:[EMAIL PROTECTED];tag=000cce3a7be800087fd8099f-62cc5396 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Mon, 06 Sep 2004 10:01:57 GMT CSeq: 102 INVITE User-Agent: CSCO/7 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Content-Type: application/sdp Content-Length: 253 v=0 o=Cisco-SIPUA 23148 13380 IN IP4 131.130.220.101 s=SIP Call c=IN IP4 131.130.220.101 t=0 0 m=audio 30596 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 15 headers, 11 lines Using latest request as basis request Sending to 83.136.32.160 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 131.130.220.101:30596 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x10f(G723|GSM|ULAW|ALAW|G729A), peer - audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0x10c(ULAW|ALAW|G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x0(EMPTY) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 83.136.32.160;branch=z9hG4bK1cba.998c27f1.0 Via: SIP/2.0/UDP 131.130.220.101:5060;branch=z9hG4bK03521c86 From: Kurt Bauer sip:[EMAIL PROTECTED];tag=000cce3a7be800087fd8099f-62cc5396 To: sip:[EMAIL PROTECTED];tag=as6191c2dd Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=troubadix.univie.ac.at, nonce=5276f268 Content-Length: 0 to 83.136.32.160:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user 'kb' troubadix*CLI Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 83.136.32.160;branch=z9hG4bK1cba.998c27f1.0 From: Kurt Bauer sip:[EMAIL PROTECTED];tag=000cce3a7be800087fd8099f-62cc5396 Call-ID: [EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as6191c2dd CSeq: 102 ACK User-Agent: Sip EXpress router(0.8.12-tcp_nonb-tls (i386/linux)) Content-Length: 0 8 headers, 0 lines /example sip.log --note the SIP/2.0 407 Proxy Authentication Required sip.conf ; ; SIP Configuration for Asterisk ; [general] port=5060 bindaddr=0.0.0.0 realm=troubadix.univie.ac.at disallow=all allow=ulaw allow=alaw allow=g729 allow=g723.1 allow=gsm [at43_in] type=peer host=sip.at43.at context=at43 insecure=very deny=0.0.0.0/0.0.0.0 permit=83.136.32.160/255.255.255.255 /sip.conf ___ 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] Going to voicemail instead of queue if no agent is logged in ?
Hi,
I did this the following way:
-) define a global variable - AGENTS_AVAIL=0
-) when agent logs in increment -
SetGlobalVar(AGENTS_AVAIL=$[${AGENTS_AVAIL} + 1]);
-) when agent logs off decrement -
SetGlobalVar(AGENTS_AVAIL=$[${AGENTS_AVAIL} - 1]);
-) when queue is called evaluate and goto label -
gotoif,$[${AGENTS_AVAIL}]?${Q}:${NO_Q)
Hope that helps and if there is an easier way of doing this please show me
how.
br,
Kurt
--On Tuesday, August 31, 2004 09:57:29 PM +0200 Robert Rozman
[EMAIL PROTECTED] wrote:
Hi,
I'd like to implement scenario to send user to operator's queue by default
(if doesn't dial any extension) but only if there is operator agent
logged, so user could get response. If not I'd like to send it to
voicemail...
Any quick advice ?
Thanks in advance,
Robert.
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Re: [Asterisk-Users] strange problem PBX-Asterisk
Thanks for the hints, 'overlapdial=yes' did the trick. br, kurt --On Tuesday, August 24, 2004 10:08:08 PM +0200 Peter Svensson [EMAIL PROTECTED] wrote: On Tue, 24 Aug 2004, Christian Victor wrote: maybe I oversee somth. very obvious, but I'm a little puzzled about the following 'error': When I make a call from the PBX to * I get number not available, but on debug I see, that asterisk is searching just for the first digit in the extension, which of course doesn't exist, eg: I seems that you PBX uses Overlap Dial and transmits the extensien digit by digit and Asterisk expects the extension to be en bloc. So when it receives anything from the PBX (wich is in this case the first digit) Asterisk thinks that this is the whole block of extension. Don't know how to fix it though. ;-) This in case you are on a PRI: see overlapdial at http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf It needs to be set on pri links where ovarlap dialing is used, even incoming towards asterisk. Without more information on the connections between the systems and the configuration it is hard to figure out what is wrong. Peter ___ 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] strange problem PBX-Asterisk
Hi, maybe I oversee somth. very obvious, but I'm a little puzzled about the following 'error': When I make a call from the PBX to * I get number not available, but on debug I see, that asterisk is searching just for the first digit in the extension, which of course doesn't exist, eg: I dial 77 (for conn to *) and 12345 (valid extension) on the console I see: -- Extension '1' in context 'sip-local' from '+ 14070' does not exist. Rejecting call on channel 0/1, span 1 Everything worked fine before an update on Friday, but I haven't changed any config files then. I 'downgraded' to 1.0RC2 today, but then same problem. If any of you has any hints, please let me know. Thanks a lot, br, Kurt ___ 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] Zaptel Problem after Upgrade
Hi, after upgrading to latest CVS (20.08) I have a problem with the connection to PSTN: When I try to make a call from PSTN to * I hear the number not available' sound and the following warnings on * console: Aug 20 12:09:55 WARNING[180235]: chan_zap.c:6762 zt_pri_error: PRI: !! Unknown IE 1329 (len = 3) Aug 20 12:09:55 WARNING[180235]: chan_zap.c:6762 zt_pri_error: PRI: !! Unknown IE 1330 (len = 3) The other way round still works although I also get a warning: Aug 20 12:10:39 WARNING[180235]: chan_zap.c:6762 zt_pri_error: PRI: !! Unknown IE 1330 (len = 3) I changed no config files, so that couldn't be the problem, or has anything changed since April (that was the CVS version with which it worked just fine). Thanks, best regards, Kurt ___ 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
