[Asterisk-Users] Voice Quality bad on one side of Frame Relay
Hi , Does anyone encounter this problem ? We have installed Asterisk at Site A and have 128k Frame Relay over to Site B. We are using Zyxel 2 port FXS at Site B and Linksys PAP2-NA at Site A. We are using Ulaw at Site A and G729 at Site B. When the calls are originated from Site A to Site B, party at Site A can hear Site B voice clearly and no breaking up voice. But Site B user hears Site A voice is breaking up sometimes. The Total bandwidth usage is about 30k. We have deployed the same setup to another Site , Site C with 64k Frame Relay. Same things happen. Any Comments / Ideas ? Regards, Stephen ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SIP 400 Bad Request from Cisco 7960/7940
Doug Lytle wrote:
[EMAIL PROTECTED] wrote:
On Mon, 3 Oct 2005, Corey S. McFadden wrote:
Am I just using the Set() command wrong? It seems pretty
counter-intuitive not to enclose multi-word strings in quotes but if
that's the problem let me know.
Yeah, that's the problem.
Steve
In my case, I'm not using quotes:
exten = s,3,Set(CALLERID(Name)=${CALLERID})
exten = s,4,Set(CALLERID(Number)=${CALLERIDNUM})
Still waiting for a SIP debug with a bad request reply...
/O :-)
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[Asterisk-Users] Asterisk and NAT
Hey guys. I have to put my * behind a Firewall through nat on the firewall. The asterisk is running, but for example a register to an outside PSTN provider won't work. I enabled nat for the register but i only get Code 120 Send request. The other problem is, when i try to register with a sip phone which is behind a nat router i cant register. When the * is in official net all is working! Regards Rene ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Outgoing busy
I have a problem. Incoming calls work without problem but I cant call out. Using AAH.Gets a busy tone Anyone who can see a mistake in Outgoing settings context=from-pstn host=ipkund1.rixtelecom.se insecure=very nat=yes secret=xxx type=peer username=0406082250 Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Outgoing busy
Hi, Outgoing setting is in zapata.conf. I think you should read the wiki more ;). If what you mean by outgoing is another sip extension then you should look for extension.conf. Links: http://www.voip-info.org/wiki-Asterisk+config+zapata.conf http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf Regards, Stevanus Anders Svensson wrote: I have a problem. Incoming calls work without problem but I cant call out. Using AAH.Gets a busy tone Anyone who can see a mistake in Outgoing settings context=from-pstn host=ipkund1.rixtelecom.se insecure=very nat=yes secret=xxx type=peer username=0406082250 Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] UPDATE - 512 Calls w/ Dig Rec: NFS Setup and Benchmarks
Dear Matt, Thanks for your great work and the effort documenting the whole process. I'm sure the whole Asterisk community benefits from this kind of work and it's really something to end up in the wiki. Thumbs up! Best regards, Vahan Matt Roth wrote: List members, My previous post SUCCESS - 512 Simultaneous Calls with Digital Recording documents using a RAM disk to eliminate the I/O bottleneck associated with digitally recording calls via the Monitor application. By recording directly to a RAM disk I was able to maintain good call quality on 512 simultaneous calls. [snip] begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] sip phones on x86_64
On Tuesday 04 October 2005 00:42, Rajesh kumar wrote: I am using Kphone which works great for my purposes! You can look at twinklephone as well at http://www.twinklephone.com/ Hi, thanks all for the info, kphone does really wierd stuff and I can't get twinkle to compile. I'm looking into that gnomeeting CVS idea. -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] zttest - 100% ?
do you think it would make any difference to change the process-priority if zttest is the only running process except ssh-daemon and the login-shells ? [EMAIL PROTECTED] wrote on 30.09.2005 18:11:47: Are you starting Asterisk with the -p option (high priority?) Also, do you get a different value if you run zttest this way: nice -n -20 zttest Carlos On 9/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Digium itself is saying their cards may work not properly with zttest results below 99,98 the card itself is working the way that we can call out and receive calls, but we encountered massive echo-problems - sometimes more, sometimes less even on lines within the same phone-provider and be sure that we've been messing around with all other possible parameters for weeks without any result. Until now we've got a setup that we can live with at least until we get different hardware. It's really worse calling someone and missing the name the called person said then picking up the phone in cause of echo-cancelling parameters or even think the line is dead, or if you've got massive echoes and it takes about 30 seconds to filter them out if at all. Dirk [EMAIL PROTECTED] wrote on 30.09.2005 16:34:18: [EMAIL PROTECTED] wrote: just as an (bad) example: we are using an x206 and couldn't get the zttest above 99.975 equal what we were doing single irq, w/o acpi, w/o apic, different kernels, w/o hyperthreading, different slots, a.s.o. for an Digium wildcard TE110P so if someone got such a board to zttest 100% maybe could give some information if where's something special to run asterisk on such boards... otherwise I think there are production differences on the ibm-mainboards or the used chipsets we'll change hardware next... You don't have to have 100% on zttest. You probably won't get it. I get the same results on one of my servers and it runs perfectly. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 and NAT
René Enskat [Teamware GmbH] wrote: Hey guys. I have to put my * behind a Firewall through nat on the firewall. The asterisk is running, but for example a register to an outside PSTN provider won't work. I enabled nat for the register but i only get Code 120 Send request. The other problem is, when i try to register with a sip phone which is behind a nat router i cant register. When the * is in official net all is working! There are many, many mails and webpages out there that explain the kind of trouble you have, one of the most common. So please try voip-info.org and the mailing list archives and you'll find an answer. If you still can't get it to work, please come back to the mailing list. Try searching asterisk sip nat. /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Outgoing busy
Anders Svensson wrote: I have a problem. Incoming calls work without problem but I cant call out. Using AAH.Gets a busy tone Anyone who can see a mistake in Outgoing settings context=from-pstn host=ipkund1.rixtelecom.se insecure=very nat=yes secret=xxx type=peer username=0406082250 username is one of the most misunderstood settings in sip.conf and it's really a bad, bad, bad name. You want to set fromuser and fromdomain together with username. username has many uses, which is bad: * One is to set a default user name that is used in combination with default IP when we have no registration from a local peer. * The other use is what you are trying to set up: to set authentication username when we register and place calls to an outbound service provider. This is always used in combination with fromuser and fromdomain. The nat=yes setting seems redundant, it should be done on Rix telecom's side. When you set nat=yes you tell asterisk that the *other end* is behind nat. I do not believe a service provider run a service behind NAT. Good luck! Lycka till! /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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-0.3.5
Hi Giordano, pls. check the following things: - edit your /etc/capi.conf (or /etc/isdn/capi.conf) and adjust the settings according to your card(s) - call "capiinit" - check the status by calling "capiinfo". The output should show the details of your card(s) - if you're running asterisk as non-root: check the permissions of /dev/capi20, the user running asterisk has to have rw permissions on that device. - adjust your config in /etc/asterisk/capi.conf After that, chan_capi.so shouldbe loaded succesfully. Hope it helps cheers Jörg From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano GrandisSent: Friday, September 30, 2005 3:00 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: R: [Asterisk-Users] chan_capi-0.3.5 Thanks Jorg, its worked, but what modules i need to use it with asterisk? I insert load = chan_capi.so in /etc/asterisk/modules.conf and chan_capi.so=yes under [globals] section. When asterisk start, I get this error: == Parsing '/etc/asterisk/modules.conf': Found [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Sep 30 16:00:06 WARNING[8294]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Sep 30 16:00:06 WARNING[8294]: chan_capi.c:2812 unload_module: Unable to unregister from CAPI! == Unregistered channel type 'CAPI' Sep 30 16:00:06 WARNING[8294]: loader.c:391 load_modules: Loading module chan_capi.so failed! Thanks again! Giordano Grandis g.grand[EMAIL PROTECTED] Le informazioni contenute nella presente e-mail e nei documenti eventualmente allegati possono essere confidenziali e sono comunque riservate al destinatario della stessa. La loro diffusione, distribuzione e/o copiatura da parte di terzi è proibita. Se avete ricevuto questa comunicazione per errore, Vi preghiamo di informare immediatamente il mittente del messaggio e di distruggere questa e-mail. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Per conto di Jörg WolfInviato: venerdì 30 settembre 2005 14.15A: Asterisk Users Mailing List - Non-Commercial DiscussionOggetto: RE: [Asterisk-Users] chan_capi-0.3.5 Giordano, you simply don't have capi installed... On debian sarge you can install the following packages: - capiutils - libcapi20-dev Hope it helps cheers Jörg From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Giordano GrandisSent: Friday, September 30, 2005 1:37 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] chan_capi-0.3.5 Hi all, im tryinf to install chan_capi but i get this error [EMAIL PROTECTED]:/usr/src/chan_capi-0.3.5# make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DNEVER_EVER_EARLY_B3_CONNECTS -DCAPI_ES -DCAPI_GAIN -DDEFLECT_ON_CIRCUITBUSY -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c chan_capi.c:36:20: capi20.h: No such file or directory In file included from chan_capi.c:39 Anyone cha help me? Thanks Giordano ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Outgoing busy
stevanus wrote: Hi, Outgoing setting is in zapata.conf. I think you should read the wiki more ;). If what you mean by outgoing is another sip extension then you should look for extension.conf. Links: http://www.voip-info.org/wiki-Asterisk+config+zapata.conf http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf And in fact it was all about sip.conf ;-) As you say, connecting to SIP service providers is well documented on the wiki, but not on those pages. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Re: FreeTDS 0.63
In article [EMAIL PROTECTED], Richard Cook [EMAIL PROTECTED] wrote: I thought maybe someone was using 0.63 with code they developed themselves. The TDS API has been un-published in 0.63: one is expected only to use dblib or ctlib, neither of which has any support in Asterisk. For 0.63 you have to use the unixODBC layer on top. See README.tds in the Asterisk doc directory. Where do you find 0.62.x? http://ibiblio.org/pub/Linux/ALPHA/freetds/old/ BTW, please try to post in plain text, not HTML. Thanks! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk forwarding SIP with Remote-Party-ID
I'm finding that I'm a bit disappointed that Asterisk doesn't naturally forward the Remote-Party-ID from inbound SIP calls (where trustedrpid=yes) to outbound SIP calls. I guess this is going to be something we have to use SER for, unless we make our own custom build (which I'm reluctant to do). Is there a good reason that the main version doesn't do it? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Dial pattern sort order
Hi! Is there a simple way for an * newbie to force * to use different sip-trunks for different calls. I have 2 siptrunks, one for inland calls and one for international calls. All in country numbers starts with 0 and all international starts with 00. This I have configured in the outbound routing. But * always use the incountry trunk because the 0. dialpattern is also true for international calls How to fix this? Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 and NAT
You've not said much about your firewall setup. I presume you've opened up 5060 and RTP ports? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
AW: [Asterisk-Users] Asterisk and NAT
Hi, Yes i opened 5060 and range -20001 The firewall is not blocking. I tried to set the externip and localnet but can't register to the pstn gateway and can't onnect with my nat phones. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Alex Lake Gesendet: Dienstag, 4. Oktober 2005 11:47 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] Asterisk and NAT You've not said much about your firewall setup. I presume you've opened up 5060 and RTP ports? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] sip phones on x86_64
me too looking for softphone...not able to enable kphone Can anyone please highlight more on it. ThX /Gurmi On 10/4/05, Wayne Gemmell [EMAIL PROTECTED] wrote: On Tuesday 04 October 2005 00:42, Rajesh kumar wrote: I am using Kphone which works great for my purposes! You can look at twinklephone as well at http://www.twinklephone.com/ Hi, thanks all for the info, kphone does really wierd stuff and I can't get twinkle to compile. I'm looking into that gnomeeting CVS idea. -- Regards Wayne Gemmell Tel Fax: (011) 894-4081 Cell : 072 836 4325 Email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 forwarding SIP with Remote-Party-ID
Alex Lake wrote: I'm finding that I'm a bit disappointed that Asterisk doesn't naturally forward the Remote-Party-ID from inbound SIP calls (where trustedrpid=yes) to outbound SIP calls. I guess this is going to be something we have to use SER for, unless we make our own custom build (which I'm reluctant to do). Is there a good reason that the main version doesn't do it? RPID support was recently added to CVS head. Please check that and see whether it does what you want to do or not. Thank you for your time testing this. Regards, /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Dial pattern sort order
Anders Svensson wrote: Hi! Is there a simple way for an * newbie to force * to use different sip-trunks for different calls. I have 2 siptrunks, one for inland calls and one for international calls. All in country numbers starts with 0 and all international starts with 00. This I have configured in the outbound routing. But * always use the incountry trunk because the 0. dialpattern is also true for international calls Anders, read the extensions.conf sample file that was installed either in your source directory or in the /etc/asterisk directory if you ran make configs after installation. We have pattern matching that matches 0-9, 1-9 and 2-9 so you can easily make this work properly for you. We haven't got a full set of documentation for Asterisk within the source, but please read the ones we have! There are plenty of advice in the sample configuration files and a lot of information in the /doc directory. Lycka till! /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk as H323 gateway
Is there anyone who is currently using Asterisk as a production H323 gateway? And using which combination of asterisk and H323 (chan_h323, chan_oh323?) The main issue is interoperability with other H323 parties (Cisco AS53xx, Nextone, etc). Searching the mailing list it seems that both h323 and oh323 are not so stable, is it only an impression or using h323 is really not so advisable? Francesco Pellegrini [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] What does the error stale nonce' mean?
On 10/3/05, Morten Isaksen [EMAIL PROTECTED] wrote: On 10/3/05, Olle E. Johansson [EMAIL PROTECTED] wrote: Does anyone know what stale nonce is?I've answered this question many times, so you should be able to find the answer...A stale nonce is when a device tries to re-authenticate with a noncethat is no longer valid. We are telling them that the nonce they used isinvalid, and re-issue a new challenge and a fresh nonce. It's just an informative message, that I propably should move away to a debug levelof some kind. I get this error when I use a Audiocodes MP-124 against Asterisk 1.2beta1 and asterisk refuses the call. When I useCVS-D2005.02.12.14.37.11-04/13/05-16:14:03 it works fine. I do not have access to the debug and log file now, but I will send them tomorrow. Here is the output from sip debug. I hope someone can explain what is wrong. -- SIP read from 10.131.2.1:5060:INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0Via: SIP/2.0/UDP 10.131.2.1 ;branch=z9hG4bKaciipncbQMax-Forwards: 70From: sip:[EMAIL PROTECTED];tag=1c1850211233To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED]CSeq: 1 INVITEContact: sip:[EMAIL PROTECTED]Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATEUser-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006Content-Type: application/sdpContent-Length: 242 v=0o=AudiocodesGW 644554 101011 IN IP4 10.131.2.1s=Phone-Callc=IN IP4 10.131.2.1t=0 0m=audio 6070 RTP/AVP 8 0 96a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000a=rtpmap:96 telephone-event/8000a=fmtp:96 0-15a=ptime:20a=sendrecv --- (13 headers 12 lines)---Using INVITE request as basis request - [EMAIL PROTECTED]Sending to 10.131.2.1 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.131.2.1:5060:SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKaciipncbQ From: sip:[EMAIL PROTECTED];tag=1c1850211233To: sip:[EMAIL PROTECTED];user=phone;tag=as6a339401Call-ID: [EMAIL PROTECTED]CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFYContact: sip:[EMAIL PROTECTED]Proxy-Authenticate: Digest realm=asterisk, nonce=22a96479 Content-Length: 0 ---Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 msFound user '070001'localhost*CLI-- SIP read from 10.131.2.1:5060:ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKaciipncbQMax-Forwards: 70From: sip:[EMAIL PROTECTED];tag=1c1850211233To: sip:[EMAIL PROTECTED];user=phone;tag=as6a339401Call-ID: [EMAIL PROTECTED]CSeq: 1 ACKContact: sip:[EMAIL PROTECTED]Supported: em,timer,replaces,pathAllow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006Content-Length: 0 --- (12 headers 0 lines)---localhost*CLI-- SIP read from 10.131.2.1:5060:INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKaclMBIpvuMax-Forwards: 70From: sip:[EMAIL PROTECTED];tag=1c1850211233To: sip:[EMAIL PROTECTED];user=phoneCall-ID: [EMAIL PROTECTED]CSeq: 2 INVITEProxy-Authorization: Digest username=070001,realm=asterisk,nonce=22a96479 ,uri= sip:[EMAIL PROTECTED],algorithm=MD5,response=41cc6e74fc333e770fa28a7db158a495Contact: sip:[EMAIL PROTECTED]Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATEUser-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006Content-Type: application/sdpContent-Length: 242 v=0o=AudiocodesGW 644554 101011 IN IP4 10.131.2.1s=Phone-Callc=IN IP4 10.131.2.1t=0 0m=audio 6070 RTP/AVP 8 0 96a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000a=rtpmap:96 telephone-event/8000a=fmtp:96 0-15a=ptime:20a=sendrecv --- (14 headers 12 lines)---Using INVITE request as basis request - [EMAIL PROTECTED]Sending to 10.131.2.1 : 5060 (non-NAT) Oct 4 13:20:51 NOTICE[4078]: chan_sip.c:5710 check_auth: stale nonce received from 'sip:[EMAIL PROTECTED];user=phone'Reliably Transmitting (no NAT) to 10.131.2.1:5060:SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKaclMBIpvuFrom: sip:[EMAIL PROTECTED] ;tag=1c1850211233To: sip:[EMAIL PROTECTED];user=phone;tag=as6a339401Call-ID: [EMAIL PROTECTED]CSeq: 2 INVITE User-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFYContact: sip:[EMAIL PROTECTED]Proxy-Authenticate: Digest realm=asterisk, nonce=0e317db4 Content-Length: 0 ---Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 msFound user '070001'localhost*CLI-- SIP read from 10.131.2.1:5060:ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0Via: SIP/2.0/UDP 10.131.2.1;branch=z9hG4bKaclMBIpvuMax-Forwards: 70From: sip:[EMAIL PROTECTED];tag=1c1850211233To: sip:[EMAIL PROTECTED];user=phone;tag=as6a339401Call-ID: [EMAIL PROTECTED]CSeq: 2 ACKContact: sip:[EMAIL PROTECTED]Supported: em,timer,replaces,pathAllow:
[Asterisk-Users] CallerID octoBRI connected on voxtream parlay i60
We have parlay i60 LCR connected to octoBRI card on i60 bri ports. When i call a number on asterisk and call is connected to i60 over BRI ports and call goes to PRI line callerid is presented without last 2 digits. greetings Milos ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] DIAX not working properly
What is the version of Diax ? regards Pierre 2005/10/4, amna saleem [EMAIL PROTECTED]: Hi! I am facing some problems with my asterisk-1.0.3.I am using DIAX phones as clients ,but sometimes they donot register with the asterisk server.Also if I don`t restart my asterisk frequently the registration of DIAX phones expires. Anyone who can help me please reply Regards, Amna ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] [Fwd: Eicon Diva 2.01 S/T PCI quality problems]
Found out what's wrong, maybe it can help others.. my linux's package manager automatically pulled the most fresh version of asterisk, 1.2beta. Downgrading to 1.0.8 solved all problems described below, I get excellent quality and no noise now. cheers, Kristof Kristof Jozsa wrote: Hi all, I'm experimenting with Asterisk on linux using an Eicon Diva 2.01 S/T PCI card. I'd set up the card using the hisax driver and isdn4linux (titled as Old ISDN4Linux (obsolete) in the 2.6.12 kernel. I can make SIP calls and outgoing phone calls as well, but gathered a few problems on my way: 1. Plain SIP calls using softphones on windows clients work fine not counting the delay I'm experimenting. We talk about 1-1.5 secs delay in the speech which is rather distrubing (no noise in the line though). 2. Outgoing calls eg. to my mobile phone has some more serious problems. Speech quality on my mobile phone is excellent. However, sound quality on the asterisk console machine where I dialled from is about unacceptable. It has about 90% static noise and about 10% speech somewhere in the middle. I have the same experience calling from a windows softphone through asterisk to my handy, maybe a little less noise (around 80%). So my questions would be: can I do anything with the issues described above? Is it a hardware problem (eg. I need to replace the old and cheap card to a more modern one)? Maybe it's a driver problem (eg. the hisax driver is known to work only with such extreme static noise)? I also don't really understand why the static appears only on the server side and not on the called side. Any help or suggestions are much appreciated, thanks very much in advance. Kristof Jozsa ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 phones?
On Mon, 03 Oct 2005 22:09:23 -0600, Stephen Bosch wrote: Hi, everyone: I'm in the processing of deciding what IP phones we should use with our Asterisk server, and I wanted to get feedback from the user community on the quality, reliability and ease of operation of Snom phones. What do you have to say about these phones? Are there other phones you'd suggest along with or instead of Snom? I used to have some Snom 200's. They were nice and with the right firmware worked well. Others can give you more specifics. I eventually shifted to Polycom 500s and 600s. These are excellent phones. They just feel great. First rate hardware and unmatched speakerphone capability. They are a bit harder to get setup as you really need to use an FTP/TFTP/HTTP server to provision effectively. The LCD display on the 600 is excellent...not quite as nice on the 500. Most recently I've added an Aastra 480 CTi in order to try the cordless extension. This is also a superb phone. The only thing that I've tried that comes close to the Polycoms. They have backlit displays which is something that I wish more manufacturers would consider. By extension I expect that the 480i is also great...just lacking the cordless extension. For my purposes in a SOHO situation the 480 CTI is a far better device than a wifi sip phone. I have a Hitachi WIP-5000 that I will soon resell. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Quad PRI Problems
I have been getting quite a bit of PRI Resets using my Quad PRI Digium card. Prior to the resets I am getting similar notices to the following chan_zap.c:7482 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 Telco claims the PRI's are fine on their end and that it is my unit. Is this timing? (google somewhat leads to this) I am running 1.08 asterisk zaptel libpri. Any help would be greatly appreciated. ~ron Zaptel.conf span=1,1,0,esf,b8zs # connects to an Adtran FXS TA624 em=1-24 span=2,1,0,esf,b8zs # Connects to Bell Company 1 bchan=25-47 dchan= 48 span=3,1,0,esf,b8zs # Connects to Bell Company #2 bchan=49-71 dchan= 72 span=4,1,0,esf,b8zs # Connects to Brook Trout CArd em=1-4 defaultzone=us loadzone=us [channels] context=from-internal-receiver ; Points to the default context of your extensions.conf language=en faxdetect=none usecallerid=yes callerid=asreceived threewaycalling=yes transfer=yes signalling=featd ; FXS for ringing phones group=0 flash=350 rxwink=300 prewink=20~~ echocancel=no ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=no immediate=no channel = 1-24 signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national pridialplan=unknown echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=no echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds group=1 context=from-pstn channel = 25-47 ; Set this to 1-15,17-31 for E1 group=2 signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national channel = 49-71 ; Set this to 1-15,17-31 for E1 group=3 signaling=em_w channel = 73-76 oledata.mso Description: Binary data ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Quad PRI Problems
You can't put four span in timing, because only one must be like nmaster sincronization. If one of your telco provide time for your card. Put second value in all span to 0. regards, srsergio -Mensaje original- De: Ronald Hartmann [mailto:[EMAIL PROTECTED] Enviado el: martes, 04 de octubre de 2005 14:33 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] Quad PRI Problems I have been getting quite a bit of PRI Resets using my Quad PRI Digium card. Prior to the resets I am getting similar notices to the following chan_zap.c:7482 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 Telco claims the PRI's are fine on their end and that it is my unit. Is this timing? (google somewhat leads to this) I am running 1.08 asterisk zaptel libpri. Any help would be greatly appreciated. ~ron Zaptel.conf span=1,1,0,esf,b8zs # connects to an Adtran FXS TA624 em=1-24 span=2,1,0,esf,b8zs # Connects to Bell Company 1 bchan=25-47 dchan= 48 span=3,1,0,esf,b8zs # Connects to Bell Company #2 bchan=49-71 dchan= 72 span=4,1,0,esf,b8zs # Connects to Brook Trout CArd em=1-4 defaultzone=us loadzone=us [channels] context=from-internal-receiver ; Points to the default context of your extensions.conf language=en faxdetect=none usecallerid=yes callerid=asreceived threewaycalling=yes transfer=yes signalling=featd ; FXS for ringing phones group=0 flash=350 rxwink=300 prewink=20~~ echocancel=no ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=no immediate=no channel = 1-24 signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national pridialplan=unknown echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=no echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds group=1 context=from-pstn channel = 25-47 ; Set this to 1-15,17-31 for E1 group=2 signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national channel = 49-71 ; Set this to 1-15,17-31 for E1 group=3 signaling=em_w channel = 73-76 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Three-way calling over SIP and IAX using softphone
Hi guys, Does anyone know of a way where I can bring a third person in on my conversation. Say I'm on a IAX or SIP call from a softphone DIAX or IAXCOMM and am speaking to someone now I want to quickly bring another SIP or IAX extension into this call so the three of us can speak to each other. I know I could do this by transfering the first person into a meetme then calling the second person and transfering him into the same meetme but thats too much trouble then I have to transfer myself into that same conference thats too much trouble can I do the same quicker and hopefully without meetme. I just want to be able to talk to two other people at the same time. Thanks -- regards Vikram ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Monitor in AGI
Hello Does anyone have an example of how to use the MONITOR command from an AGI-script ? I have tried different methods, but none of them worked :-( I'm using Python MIR ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Voice Quality bad on one side of Frame Relay
On Tue, 04 Oct 2005 14:28:47 +0800, Stephen wrote: Hi , Does anyone encounter this problem ? We have installed Asterisk at Site A and have 128k Frame Relay over to Site B. We are using Zyxel 2 port FXS at Site B and Linksys PAP2-NA at Site A. We are using Ulaw at Site A and G729 at Site B. When the calls are originated from Site A to Site B, party at Site A can hear Site B voice clearly and no breaking up voice. But Site B user hears Site A voice is breaking up sometimes. The Total bandwidth usage is about 30k. We have deployed the same setup to another Site , Site C with 64k Frame Relay. Same things happen. It sounds like you're very bandwidth constrained. A ULAW leg is going to be 64k, or actually around 80k with IP overhead. Not sure abour FR overhead. I can't see how you'll get anythin at site C without using a compressed codec. G729 is a good choice for codec to get over the bandwidth issue. Why not use it in all cases? You can always experiment with GSM for no cost. Also, is there any other network activity beyond the voip streams? In such a bandwidth limited instalation QoS management is going to be critical. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Quad PRI Problems
On Tuesday 04 October 2005 08:32, Ronald Hartmann wrote: I have been getting quite a bit of PRI Resets using my Quad PRI Digium card. You've got a problematic setup for Digium's Zaptel cards. You're also running an old version of Asterisk. 1.09 is the stable release and 1.2 is the upcoming next stable release. span=1,1,0,esf,b8zs # connects to an Adtran FXS TA624 span=2,1,0,esf,b8zs # Connects to Bell Company 1 span=3,1,0,esf,b8zs # Connects to Bell Company #2 span=4,1,0,esf,b8zs # Connects to Brook Trout CArd First of all you've got all four spans attempting to synchronize with the other side of the respective span. Digium cards cannot do this; all four spans must share one common clock. Seeing as you have two different telcos coming in to this card, it could be difficult to achieve sync, but telcos generally all come back to one common clock anyway so it may work out all right. I'd set span 1, and 4 to a clocking value of '0' which means do not attempt to sync to this span, use the internal clock. Leave span 2 as is, with clocking set to '1', which means this span is my primary sync source. Set span 3 to a clock value of '2', which means this span is my secondary sync source. To summarize the clocking values: 0 = do not attempt to use this span for a clock source. 1 = this is my primary sync source. Synchronize the internal clock to the recovered clock from this span. 2 = this is my secondary sync source. Use this span's recovered clock as a sync source if my primary sync source is down. 3 = this is my tertiary sync source. Use this span's recovered clock as a sync source if my primary and secondary sync sources are down. 4 = ... you get the idea. Basically the setup I suggested for you is to use telco #1 as the primary sync source, and to fall back to telco #2's clock if telco #1's span goes down. I have suggested not attempting to recover clock from the Adtran nor the Brook Trout spans. Good luck. :-) -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] TDM400P recognised as Network controller: Unknown device
On Tuesday 04 Oct 2005 05:17, [EMAIL PROTECTED] wrote: On Mon, 3 Oct 2005, Aryanto Rachmad wrote: I sent an email to Digium support and got only a reply like this: Although the card is being shown as an 'Unknown Device', it should still work properly. To be honest, I am not happy with that answer. Well, does it work properly? It will. What is Digium supposed to do about lspci which isn't their code. Its lspci which does or doesn't come up with a name for the card. From the pci.ids file. Digium should email the details to Martin. # # List of PCI ID's # # Maintained by Martin Mares [EMAIL PROTECTED] and other volunteers from the # Linux PCI ID's Project at http://pciids.sf.net/. New data are always # welcome (if they are accurate), we're eagerly expecting new entries, # so if you have anything to contribute, please visit the home page or # send a diff -u against the most recent pci.ids to [EMAIL PROTECTED] # # Daily snapshot on Tue 2005-02-08 11:00:09 # ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 1.0.8 and TDM stop acking inbound calls?
Patrick Friedel wrote: Rich Adamson wrote: My office has been running Asterisk 1.0.8 and a TDM04B for a few months now without too much trouble. After a while we discovered that after a certain period (about a month), asterisk stopped acknowledging inbound calls. Since I was out of the office the first time it happened, another admin rebooted the whole box which solved the problem. The second time it happened I discovered that just restarting gracefully solved the problem, so I put that into my cron and forgot about it. (I know, it's not right, but debugging something that happens unpredictably once a month could go on for way too long to be acceptable..) Check the revision of the TDM card. If rev E/F, call digium support to get it replaced. Known problem with early versions of the card. The rev is labelled on the itty bitty xilinx chip and not under the modules, right? Dang, rev F. Okiedoke, off to digium I go. Thanks! ___ we also experienced this with asterisk 1.0.9 and rev H of the tdm with 4 fxo modules we were restarting asterisk every night via cron and this still happened in our case, 3 out of 4 fxo modules (2,3,4) crapped out and stopped ack'ing incoming calls (outgoing calls were fine) it took a reboot of the server to get the card operational again and answering calls Your problem might be with the rev of the card, but I believe it is in the zaptel software for the newer versions of asterisk. in the past, It was the zaptel software that would not restart the tdm card (400b) on a soft reboot, on some cards (rev H). Needed to hard boot to get things working, until the code was changed in wctdm.c specific for rev H. I have swapped out the tdm and modules with albeit same rev parts, but newer units to see if the issue is the same. I still think it is in the software as we did not have this issue with earlier versions (1.02) of * Please post your resolution if any. Best Regards Greg Cirino Spam and Virus Free Email included with every email account Cirelle Enterprises Inc. 25 Indian Rock Rd #421 Windham NH, 03087 603-425-2221 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] UPDATE - 512 Calls w/ Dig Rec: NFS Setup and Benchmarks
On 3 Oct 2005, at 22:54, Matt Roth wrote: List members, It has been a while, but I once implemented a simple shared database over NFS, so dredging my memory produced the following: Future Plans and Unresolved Issues == I wrote Windows software for another project that mixes leg files, indexes them by call time, and archives them after a given period of time. I plan to port that code to a set of shell scripts that will be run on the Digital Recording server out of cron. If anyone knows of an existing project that has accomplished this already, please let me know. Before writing these scripts, I have two questions that need answered: 1) How can I tell when a file is complete on the NFS server? If I recall right, you can't (not on the nfs server end). The way I used to handle this was to have the creating client rename the file once it has finished with it. (remove a leading dot is good). I think you can assume (with a decent NFS implementation) that the rename won't happen untill all the queued writes+close have occurred. 2) What will happen on the NFS client if the NFS server crashes (I expect the leg files to be written to the local mount point until the mount is reesablished)? Nothing so tidy, certainly not on files that were open at the time of the crash. To get that behavior even for new files you would need to un-mount the nfs filesystem on the client whenever there is a crash. (Hmm, kinda like the translucent filesystem...) Tim. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] TDM versions question
I have just realized while trying to research asterisk not acking incoming calls that the tdm400b card is stamped rev H, but when I issue the zap show status command in the manager interface, it indicates Wildcard TDM400P REV E/F Board 1 Which do I believe?? -- Best Regards Greg Cirino Spam and Virus Free Email included with every email account Cirelle Enterprises Inc. 25 Indian Rock Rd #421 Windham NH, 03087 603-425-2221 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Fw: [Asterisk-Users] trunking IAX2
Hello, Would like to use IAX /IAX2 to transport 30 channels inter Asterisk. I don't have any experience with that, so can someone help ?? How much bw do I need for simultaneous calls and is there any latency for SIP G711 to IAX2 and vice-versa , ... etc ? Thanks in advance for any info, Geo ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] TDM400P recognised as Network controller: Unknown device
Bob Goddard wrote: From the pci.ids file. Digium should email the details to Martin. We are well aware of that. A quick scan of that file will show that we already have the IDs for the dual/quad-span cards in the master database. The issue with the TDM400P and the single-span cards is that we are not the 'owner' the PCI vendor/device IDs on those boards, and they are not unique to our boards. The PCI subvendor/device IDs on the boards are also not consistent enough for listing in the PCI ID database, and the PCI subvendor ID is not our ID, so we cannot legitimately update that part of the database. This has -zero- effect on operation; it is purely cosmetic. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 911 Q
Currently we are working with Telco Providers to provide 911 and e911 with all the bells and whistles, including CNMAN features. This will enable you to deliver 911 calls to PSAP with out having to tell them your location. Get ready to manage DB ... check out REDSKY software -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joel Newkirk Sent: Monday, October 03, 2005 2:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 911 Q Thank you - while not directly an answer to my question, it directly addresses the root of my question, pointing me where I'll need to go to dig deeper. It also tells me what we didn't want to hear, that there's a very good possibility that we simply won't be able to ensure that the 911 call center can tell which unit a call comes from without verbal specification from the caller. j On Sun, 2005-10-02 at 08:13 -0600, Rich Adamson wrote: Asterisk is more then capable of sending the appropriate callerid info to any remote site including 911 centers. However, there is a telco between asterisk and the 911 center that may not have realistic policies/systems to accept and forward that callerid. So, your objective becomes one of what the telco will allow you to do (and their support of your objective). As one example only, the telco might have a switch that does not have PRI capabilities (I know of many of these), but they provide ANI info to the 911 centers since that _might_ be the only data they can provide. If that were the case in your environment, it doesn't make any difference what you do with asterisk, it won't be supported. I know from practical experience that a telco's switch (in most cases) will accept calleridnum via a PRI, but on most central office switches its an option that needs to be turned on. (Local telco policy _might_ say they will never do that.) Once that option is turned on, you can send almost anything to them in the form of calleridnum. The callerid name is a different story. The central office switch that _terminates_ any call (including 911 calls) will have a mechanism to do a database lookup/dip, and if that database has not been populated with an appropriate callerid name, will not provide callerid names to the 911 center (or anyone else). That essentially says you can program asterisk to send anything that you want from a callerid name perspective and it will be ignored in the US. In very general terms, only telco personnal have the access to update the callerid database, and usually that is limited to the CO prefixes they support. Also keep in mind that not all 911 centers are the same from a technical perspective. They certainly accept callerid numbers, but they may have their own local database for names (etc), or, they may also do a database lookup from some distant database. If you think about those customers that subscribe to callerid blocking, cell phones gps, and the requirements of 911 centers, its not hard to visualize several different 911 implementation approaches. Talk to a knowledgable telco person (might be somewhat difficult to find the appropriate person), and talk to the 911 center manager to better understand what options you might have available. I'd start with the 911 manager as he will know a telco person that understands the technical requirements. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] TDM versions question
Cirelle Enterprises wrote: I have just realized while trying to research asterisk not acking incoming calls that the tdm400b card is stamped rev H, but when I issue the zap show status command in the manager interface, it indicates Wildcard TDM400P REV E/F Board 1 Please contact Digium Technical Support. I don't believe you should be seeing that combination. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] can't reject call using macro-screen
I'm trying to screen the call transfert to my cell phone using a exemple found on the web.
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
It work partially: while I'm prompted to accept the the caller still
her the music. But wether I accept the call or reject it I'm put in
communication with the caller.
I can see in the log that MARCO_RESULT is set to CONTINUE when I reject
the call but is doesn't hangup as documented on the wiki: CONTINUE -
Hangup the called party and continue on in the dialplan from where you
called Dial
Any hint?
exten = 447,1,Playback(pls-wait-connect-call)
exten = 447,2,Dial(IAX2/x/1514999,20,gmM(screen2))
exten = 447,3,Macro(vm,200)
exten = 447,101,Macro(vm,200,BUSY)
exten = 447,102,hangup
[macro-screen2]
exten = s,1,Playback(silence/1)
exten = s,2,Playback(custom/screen-from)
exten = s,3,Read(ACCEPT|custom/screen-accept|1||3)
exten = s,4,GotoIf($[${ACCEPT} = 1 ] ?6:5)
exten = s,5,SetVar(MACRO_RESULT=CONTINUE)
exten = s,6,NoOp
-- Marcel Ericmailto: [EMAIL PROTECTED]
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[Asterisk-Users] Error: -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from xxx.xxx.xxx.xxx
Hi, I am running asterisk on Fedora Core 3, Configured few extension, I receive frequent error message on * console as -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from xxx.xxx.xxx.xxx. This only comes from two extensions which I configured Any idea what does this error means Regards Shaikh ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Hang-up Detect - Yet Again
* answers the call, but if the incoming caller hangs up, * does not release the line. Is there a polarity reversal on hangup (those clicks you hear maybe)? If so then you may find that using the CVS-HEAD version of Asterisk will help hugely. Put hanguponpolarityswitch=yes in your zapata.conf But I'm positive that the definitive answer to most people's hang up detection problems would be some code in chan_zap to detect a tone other than busy on hangup. For example on my line is it a continuous tone. On yours you get a dial-tone. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] X100p Problem, randomly hungup pstn line
Right at the end of your Zapata.conf you have: #include zapata_additional.conf hanguponpolarityswitch ;Include genzaptelconf configs #include zapata-auto.conf Remove that hanuponpolarityswitch as you already have hanguponpolarityswitch=yes earlier on, and I don't know what having the second one, with no =yes/no would do. Then, with regards to logging, in logger.conf (and not logging.conf like I said in my original message -- but you noticed that already :-) ) Looks for the console = and messages = lines. At the moment they will be something like console = notice,warning,error and messages = notice,warning,error If you add ,debug without the quotes to either line, debug information will then be shown on the console (if you add it to the console line) or in /var/log/asterisk/messages (or somewhere similar) if you add it to the messages line. To view the contents of /var/log/asterisk/messages in real time (constantly updated), use the following command from the command line (not asterisk's command line but your Linux box's command line) tail -f /var/log/asterisk/messages tail, by itself, shows the last 5 or so lines. Tail, with -f, keeps looking, and so you get a scrolling log of what is being added. Very useful. Restart asterisk for the new logging changes to be shown. You should then be able to see some hopefully useful debug messages as your call progresses. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Dynamic feature support recently added to CVS HEAD
I've been trying to work with the dynamic feature support.. IE adding codes like *2 to features.conf that can trigger a dialplan application to run. I've been unable to get goto to work properly. AGI also seems to not function correctly if called as a feature. Anyone else playing around with this feature might have some insight? -bill [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] TDM versions question
Kevin P. Fleming wrote: Cirelle Enterprises wrote: I have just realized while trying to research asterisk not acking incoming calls that the tdm400b card is stamped rev H, but when I issue the zap show status command in the manager interface, it indicates Wildcard TDM400P REV E/F Board 1 Please contact Digium Technical Support. I don't believe you should be seeing that combination. ___ Just did, thanks Best Regards Greg Cirino Spam and Virus Free Email included with every email account Cirelle Enterprises Inc. 25 Indian Rock Rd #421 Windham NH, 03087 603-425-2221 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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: AW: [Asterisk-Users] Asterisk and NAT
I guess you could post your config files here and hope that someone feels inclined to look them over! ;-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk Calling Card Platform
Can anyone tell me if there is a Calling Card Platform in which I can use in conjuction with Asterisk that can give me Authentication via the caller id of the user. I don't want a PIN based Calling Card system, but the software to be able to recognize the caller ID information and authenticate the users and allow them to call through the asterisk box. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 Calling Card Platform
On Tue, 2005-10-04 at 08:35 -0500, [EMAIL PROTECTED] wrote: Can anyone tell me if there is a Calling Card Platform in which I can use in conjuction with Asterisk that can give me Authentication via the caller id of the user. I don't want a PIN based Calling Card system, but the software to be able to recognize the caller ID information and authenticate the users and allow them to call through the asterisk box. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 you should possibly rethink just using callerid as authentication since callerid is so easy to spoof signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Auto attendant
Hi! Where can a newbie find some info about how to set up an auto attendant extension? Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] spandsp and page orientation
I have just installed spandsp-0.0.2 onto my Asterisk 1.0.9 I am using an old Intel 536EP (actually found drivers that work) BUT...all my faxes are coming in landscape mode Has anyone come across this? any fixes? Shawn ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] IODBC instead of UNIXODBC
Hello. It's possible to use IODBC instead UNIXODBC with realtime? As I see, the res Makefile load a odbcinst.h file, but in IODBC there's not this file. I change the res Makefile (iodbcinst.h instead odbcinst.h) and the make create the res_odbc.so. But when asterisk boot it don't start showing: [res_odbc.so]Oct 4 10:24:43 WARNING[9748]: loader.c:314 __load_resource: libiodbc.so.2: cannot open shared object file: No such file or directory Oct 4 10:24:43 WARNING[9748]: loader.c:543 load_modules: Loading module res_odbc.so failed! There is something else I have do? Thanks. JS 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo Canceling
I have been battling echo since we installed a new system at one of our clients. I am using a single span digium card. I believe this is the first time someone has setup a PRI in this area (its way out in the middle of nowhere). We get slight echo on all calls, and when calling some numbers (long distance calls but still in the local area), we get very loud echo. The person calling out can hear their own voice at the same volume about a half second after they speak. Its been getting annoying and I have tweaked everything I can, and even tried diferent echo cans in zaptel. I have talked to the telco (Bellsouth) which does both local and long distance, and they couldn't help me. They can fix the long distance echo, but they seem to fix it only on a per number basis (I call with one number, and they fix it, but then I call with another number, and they fix that, but it never actually fixes ALL the echo for long distance). I am looking into hardware echo cancelers, but I haven't found a place to get one. Would switching to Sangoma cards help fix my issues? I know Digium cards have issues on some servers/motherboards, and yet I haven't heard of any issues with Sangoma cards. If someone has had issues then I would like to hear them. I know I can find some hardware echo can's on ebay and wire it up, but I would like to have a better product that comes as a single span T1. Thanks, Tad ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Two Questions
1) What do these two notices mean? Oct 4 09:34:30 NOTICE[49584]: chan_iax2.c:5777 socket_read: Rejected connect attempt from 198.22.67.70, request '[EMAIL PROTECTED]' does not exist Oct 4 09:34:51 NOTICE[49584]: chan_iax2.c:5777 socket_read: Rejected connect attempt from 66.234.228.170, request '[EMAIL PROTECTED]' does not exist 2) I have ran the registration utility and done the setup of g.729 codec. How do I enable that (settings are enabled in modules.conf) so that it will use that codec priority before ulaw/alaw, et cetera? I just have allow=g729 in the sip/iax configs Joshua __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Call-in/Call-out
Hello, How would I setup where I call into my number and press say 911 and then it would ask for a pass and would accept it and then would prompt for a number so I could call out of my number on the road? Joshua __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] UPDATE - 512 Calls w/ Dig Rec: NFS Setup and Benchmarks
On 10/3/05, Matt Roth [EMAIL PROTECTED] wrote: Before writing these scripts, I have two questions that need answered:1) How can I tell when a file is complete on the NFS server? We do something similar with a perl script, it grabs the file sizes of the 2 legs of the recording, then waits 5 seconds and checks the filesizes again, and if the recording is finished the files should not have grown in size over 5 seconds so it mixes the files together. This of course would not work if you had something like recording-pausing enabled on your server though. MATT--- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SNOM Subscribe/Notify
I'm using a SNOM 360 with Ver 4.3 software. Asterisk is Asterisk CVS-D2005.05.02.22.00.00-05/04/05 (BRI Stuff + Head) I've used the wiki info to set up some lines to monitor some internal extensions. When the extension is rung - the lamp comes on, when the call is answered, the lamp goes off.. I was expecting something a little more exciting - like the lamp to flash when the extension was ringing and for the lamp to go on when the extension was busy - either incoming or outgoing calls. Am I missing something here??? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo Canceling
On Tuesday 04 October 2005 10:37, Tad Heckaman wrote: the middle of nowhere). We get slight echo on all calls, and when calling some numbers (long distance calls but still in the local area), we get very loud echo. The person calling out can hear their own voice at the same volume about a half second after they speak. Its We get that with our Bell Canada PRI only on certain (local) exchanges. I'm not exactly sure what is causing it but it's been a problem since we installed the PRI. Would switching to Sangoma cards help fix my issues? I know Digium cards have issues on some servers/motherboards, and yet I haven't heard of any issues with Sangoma cards. If someone has had issues then I would like to hear them. I doubt it, the echo can seems to be working just fine, but in your case (and mine) it seems like it must be getting disabled. I must admit it's been a while since I've dove into this problem, but if you execute dmesg -c on the asterisk box, make a call that will echo, and execute dmesg -c again, is there any mention of echo canceller disabled on channel 5 in the resultant output? -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Seeking Asterisk Solution For mid sized corp.
Hey Guys, I have a new task to tackle. I need to make asterisk save me as much $$$ From Ma Bell As possible. Here is the Scenario. I have 50 storefronts throughout the state of Michigan. Each location has 2 voice lines in a hunt group and a FAX Line with DSL on it. I also have a large Asterisk Box here at the corporate office Currently with one PRI Terminated to it. I obviously need to keep at least one analog line on site at each location for the DSL to work. But how could I cut the other two lines. I was trying to figure out how to keep my hunt groups working though. Would I have to port my existing numbers over and have asterisk do all the hunting and forward to a new unlisted number using a 2ns PRI Channel. Or would it be easier to just use VOIP for everything and only use the Pots line to bring in the DSL. For e911 purposes I do need the stores to still be able to pick up the local POTS line and call out. The other issue is the faxes, if I could use asterisk to distribute the faxes and use voip from the stores to send them out via asterisk I could save thousands there alone. Let me know if anyone has had a similar setup. Thanks in advance!! Regards Tim King ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Number Restriction
Hello, I have a block of 25 DIDs and have 10 phones on the network. I want when a person tries to call out for * to pick a number for the CIDN and I want to make sure that the number isn't duplicated while it's in use. Joshua __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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, cisco 3640's and DIDs...
Add direct-inward-dial to your dial peer and it should work fine. -Greg On Mon, 2005-10-03 at 15:48 -0700, Tim Pozar wrote: I would think I could do this but for some reason I am stymied. I have a PRI from RCN connected to a cisco 3640 (in my day cisco is all lower case :-)). My config looks something like this on the cisco... - voice-card 3 dsp services dspfarm ! ip cef ! isdn switch-type primary-5ess ! controller T1 3/0 framing esf linecode b8zs pri-group timeslots 1-24 description RCN PRI at SF7 ! interface FastEthernet1/0 no ip address duplex auto speed auto ! interface Serial3/0:23 no ip address dialer-group 1 isdn switch-type primary-5ess isdn incoming-voice voice no cdp enable ! voice-port 3/0:23 connection plar 1000 ! dial-peer cor custom ! dial-peer voice 1 voip destination-pattern 1000 session protocol sipv2 session target ipv4:1.2.3.4:5060 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 2 pots destination-pattern 9T port 3/0:23 ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:1.2.3.5 ! - But of course with that what get's set as the DID number is 1000. I need to find out how to get the DID number passed to asterisk. Any thoughts from folks out there? Thanks... Tim PS... Here is a show ver from the 3640... vr01-200p-sfoshow ver Cisco Internetwork Operating System Software IOS (tm) 3600 Software (C3640-IS-M), Version 12.3(16), RELEASE SOFTWARE (fc4) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2005 by cisco Systems, Inc. Compiled Tue 23-Aug-05 20:03 by ssearch Image text-base: 0x60008B00, data-base: 0x61BFA000 ROM: System Bootstrap, Version 11.1(19)AA, EARLY DEPLOYMENT RELEASE SOFTWARE (fc1) ROM: 3600 Software (C3640-IS-M), Version 12.3(16), RELEASE SOFTWARE (fc4) vr01-200p-sfo uptime is 3 weeks, 3 days, 22 hours, 22 minutes System returned to ROM by reload at 00:24:09 UTC Fri Sep 9 2005 System restarted at 00:25:48 UTC Fri Sep 9 2005 System image file is flash:flash:c3640-is-mz.123-16.bin cisco 3640 (R4700) processor (revision 0x00) with 124928K/6144K bytes of memory. Processor board ID 10311643 R4700 CPU at 100MHz, Implementation 33, Rev 1.0 Bridging software. X.25 software, Version 3.0.0. SuperLAT software (copyright 1990 by Meridian Technology Corp). Primary Rate ISDN software, Version 1.1. 2 FastEthernet/IEEE 802.3 interface(s) 24 Serial network interface(s) 1 Channelized T1/PRI port(s) DRAM configuration is 64 bits wide with parity disabled. 125K bytes of non-volatile configuration memory. 32768K bytes of processor board System flash (Read/Write) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo Canceling
zaptel Disabled echo canceller because of tone (rx) on channel 16 I just did dmesg -c and thats what I got... I think there was a call allready in progress. Whats that about? On 10/4/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 04 October 2005 10:37, Tad Heckaman wrote: the middle of nowhere). We get slight echo on all calls, and when calling some numbers (long distance calls but still in the local area), we get very loud echo. The person calling out can hear their own voice at the same volume about a half second after they speak. Its We get that with our Bell Canada PRI only on certain (local) exchanges. I'm not exactly sure what is causing it but it's been a problem since we installed the PRI. Would switching to Sangoma cards help fix my issues? I know Digium cards have issues on some servers/motherboards, and yet I haven't heard of any issues with Sangoma cards. If someone has had issues then I would like to hear them. I doubt it, the echo can seems to be working just fine, but in your case (and mine) it seems like it must be getting disabled. I must admit it's been a while since I've dove into this problem, but if you execute dmesg -c on the asterisk box, make a call that will echo, and execute dmesg -c again, is there any mention of echo canceller disabled on channel 5 in the resultant output? -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Tad Heckaman ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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-in/Call-out
On Tue, 2005-10-04 at 07:44 -0700, Crystal Stream, Incorporated wrote: Hello, How would I setup where I call into my number and press say 911 and then it would ask for a pass and would accept it and then would prompt for a number so I could call out of my number on the road? Joshua show application disa -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo Canceling
After doing a quick search, it appears that maybe a bellsouth echo can is turning my echo can off? How do I tell zaptel to leave it on, regardless if it recieves that tone? On 10/4/05, Tad Heckaman [EMAIL PROTECTED] wrote: zaptel Disabled echo canceller because of tone (rx) on channel 16 I just did dmesg -c and thats what I got... I think there was a call allready in progress. Whats that about? On 10/4/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 04 October 2005 10:37, Tad Heckaman wrote: the middle of nowhere). We get slight echo on all calls, and when calling some numbers (long distance calls but still in the local area), we get very loud echo. The person calling out can hear their own voice at the same volume about a half second after they speak. Its We get that with our Bell Canada PRI only on certain (local) exchanges. I'm not exactly sure what is causing it but it's been a problem since we installed the PRI. Would switching to Sangoma cards help fix my issues? I know Digium cards have issues on some servers/motherboards, and yet I haven't heard of any issues with Sangoma cards. If someone has had issues then I would like to hear them. I doubt it, the echo can seems to be working just fine, but in your case (and mine) it seems like it must be getting disabled. I must admit it's been a while since I've dove into this problem, but if you execute dmesg -c on the asterisk box, make a call that will echo, and execute dmesg -c again, is there any mention of echo canceller disabled on channel 5 in the resultant output? -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Tad Heckaman -- Tad Heckaman ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Announcing Voice over IP Dir ectory Services (http://www.voipDS.org)
Announcing Voice over IP Directory Services (http://www.voipDS.org) Hi All, I am sure many of you are aware that by using VOIP devices one can make peer-to-peer calls. By devices I mean software phones, hardware phones and Asterisk. This feature is available, out of the box, in majority of devices today. Peer-to-peer calls are 'free' as in 'free beer' because they don't go through the service providers. All that is required to make a peer-to-peer call is 'peer connection information' of other user. To make this global, where any VOIP user could make peer-to-peer call to any other VOIP user, we need the following a central repository which stores peer connection information of all users an easy way to search and retrieve peer connection information of other users. Voice over IP Directory Services (voipDS, pronounced as 'voip' D S) precisely addresses this need. Its a central repository that stores peer connection information of all users and also provides a search mechanism by which one could search for other VOIP users. Searching and registering 'peer connection information' can be done manually via web based interface or automatically by implementing 'voipDS protocol'. For more information, please visit, http://www.voipds.org voipDS is an 'Open source' effort and all the code will be released under GPL. If you wish to take part in this effort, please visit the website and join the mailing lists. Feedback, comments and suggestions, please send to 'feedback at voipds dot org'. Thanks for your time, Balaji NJL balaji at voipDS dot org __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] IODBC instead of UNIXODBC
I would check your /etc/ld.so.conf file and make sure that you have the library path for the IODBC libraries in there... Then run ldconfig and try reloading asterisk again. Hello. It's possible to use IODBC instead UNIXODBC with realtime? As I see, the res Makefile load a odbcinst.h file, but in IODBC there's not this file. I change the res Makefile (iodbcinst.h instead odbcinst.h) and the make create the res_odbc.so. But when asterisk boot it don't start showing: [res_odbc.so]Oct 4 10:24:43 WARNING[9748]: loader.c:314 __load_resource: libiodbc.so.2: cannot open shared object file: No such file or directory Oct 4 10:24:43 WARNING[9748]: loader.c:543 load_modules: Loading module res_odbc.so failed! There is something else I have do? Thanks. JS 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo Canceling
On Tuesday 04 October 2005 10:59, Tad Heckaman wrote: zaptel Disabled echo canceller because of tone (rx) on channel 16 I just did dmesg -c and thats what I got... I think there was a call allready in progress. Whats that about? Was that after the first or second dmesg -c? Procedure: dmesg -c place call that will have terrible echo finish call dmesg -c After the 2nd dmesg -c did you see that message? I'm also assuming a relatively unloaded system, and if that on channel 16 was the channel your call went through (You'll see Zap/16-1 or something to that effect in the * CLI) then you've identified the problem. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] CVS Head - Obtaining Newbie Question
Can someone explain what is meant by CVS Head? I see references to it a lot, but don't know what it means! I am having a problem with *, and it has been suggested that installing CVS Head might help, but I don't what it is, or where to get it! TIA Leigh ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 Subscribe/Notify
Upgrade Asterisk. Versions of HEAD post 8-29-05 have this functionality built in. On 10/4/05, Mark Elkins [EMAIL PROTECTED] wrote: I'm using a SNOM 360 with Ver 4.3 software.Asterisk is Asterisk CVS-D2005.05.02.22.00.00-05/04/05(BRI Stuff + Head)I've used the wiki info to set up some lines to monitor some internalextensions.When the extension is rung - the lamp comes on, when the call isanswered, the lamp goes off..I was expecting something a little more exciting - like the lamp to flash when the extension was ringing and for the lamp to go on when theextension was busy - either incoming or outgoing calls. Am I missingsomething here???--.. ___. .__Posix Systems - Sth Africa. e.164 VOIP ready/| /| / /__ [EMAIL PROTECTED]-Mark J Elkins, Cisco CCIE/ |/ |ARK \_/ /__ LKINSTel: +27 12 807 0590Cell: +27 82 601 0496___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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-in/Call-out
How would I setup where I call into my number and
press say 911 and then it would ask for a pass and
would accept it and then would prompt for a number so
I could call out of my number on the road?
How about in whatever context you define your inbound number, add this exten:
exten = NXXNXX,1, Authenticate(911)
exten = NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]:iaxserver.com/1${EXTEN})
exten = NXXNXX,3,Hangup
-a
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[Asterisk-Users] Polycom 501: takes calls, but fast busy on dial out?
Hi, Has anyone seen this before? The phones are registered OK, and they can take incoming calls, but all I get is a fast busy whatever I dial. I've tried regular numbers, *98, etc. Looking at the Asterisk Command Line Interface, I don't see any text outputted when I try to dial out. I wonder if it's even getting to the Asterisk server. Where does it get the fast busy from--inside the phone? Another clue is the Flash Operator Panel. The extensions show up, but the devices are greyed out. Here is what sip show peers has: Name/usernameHostDyn Nat ACL Mask Port Status 15102/15102 XXX.XXX.XXX.XXX D N 255.255.255.255 1044 OK (69 ms) 15101/15101 XXX.XXX.XXX.XXX D N 255.255.255.255 1043 OK (69 ms) Do the ports matter? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo Canceling
You know what.. I have sporadic echo issues too and I just checked my dmesg and also see that! What's this all about? zaptel Disabled echo canceller because of tone (rx) on channel 3 zaptel Disabled echo canceller because of tone (rx) on channel 2 zaptel Disabled echo canceller because of tone (rx) on channel 2 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 2 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 2 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 2 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 3 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 zaptel Disabled echo canceller because of tone (rx) on channel 1 On 10/4/05, Tad Heckaman [EMAIL PROTECTED] wrote: After doing a quick search, it appears that maybe a bellsouth echo can is turning my echo can off? How do I tell zaptel to leave it on, regardless if it recieves that tone? On 10/4/05, Tad Heckaman [EMAIL PROTECTED] wrote: zaptel Disabled echo canceller because of tone (rx) on channel 16 I just did dmesg -c and thats what I got... I think there was a call allready in progress. Whats that about? On 10/4/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 04 October 2005 10:37, Tad Heckaman wrote: the middle of nowhere). We get slight echo on all calls, and when calling some numbers (long distance calls but still in the local area), we get very loud echo. The person calling out can hear their own voice at the same volume about a half second after they speak. Its We get that with our Bell Canada PRI only on certain (local) exchanges. I'm not exactly sure what is causing it but it's been a problem since we installed the PRI. Would switching to Sangoma cards help fix my issues? I know Digium cards have issues on some servers/motherboards, and yet I haven't heard of any issues with Sangoma cards. If someone has had issues then I would like to hear them. I doubt it, the echo can seems to be working just fine, but in your case (and mine) it seems like it must be getting disabled. I must admit it's been a while since I've dove into this problem, but if you execute dmesg -c on the asterisk box, make a call that will echo, and execute dmesg -c again, is there any mention of echo canceller disabled on channel 5 in the resultant output? -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Tad Heckaman -- Tad Heckaman ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo Canceling
On Tuesday 04 October 2005 11:02, Tad Heckaman wrote: After doing a quick search, it appears that maybe a bellsouth echo can is turning my echo can off? How do I tell zaptel to leave it on, regardless if it recieves that tone? The tone to disable echo cancellers is a good thing, not a bad one. Fax machines and modems tend to not work well with echo cancellation (they take care of it themselves) and if you disable the tone detection in zaptel (there's a #define for it in zconfig.h IIRC) then the chances of your faxes/modem calls working through the PRI in either direction will go down significantly. I actually have a wishlist item to enable/disable that tone detection on a per-call basis. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Announcing – Voice o ver IP Directory Services (http://www.voi pDS.org)
Balaji NJL wrote: Announcing – Voice over IP Directory Services (http://www.voipDS.org) To make this global, where any VOIP user could make peer-to-peer call to any other VOIP user, we need the following a central repository which stores peer connection information of all users an easy way to search and retrieve peer connection information of other users. Sorry to be negative, but this kind of services came up in tons when e-mail was a new service. All of the addresses registred in there is now totally un-usable because of spam. I do not really believe in global directory services as a solution... /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Can't compile ast_rxfax with Asterisk 1.2.1b
I'm trying to get ast_rxfax and ast_txfax compiling with Asterisk 1.2.1 beta. The two ast_?xfax files don't compile: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -fPIC -D_GNU_SOURCE -c -o app_rxfax.o app_rxfax.capp_rxfax.c: In function phase_e_handler:app_rxfax.c:77: warning: implicit declaration of function fax_get_transfer_statisticsapp_rxfax.c:78: warning: implicit declaration of function fax_get_far_identapp_rxfax.c:79: warning: implicit declaration of function fax_get_local_identapp_rxfax.c: In function rxfax_exec:app_rxfax.c:189: warning: pointer targets in passing argument 1 of __builtin_strncpy differ in signednessapp_rxfax.c:259: warning: passing argument 1 of fax_init from incompatible pointer typeapp_rxfax.c:260: error: t30_state_t has no member named verboseapp_rxfax.c:263: warning: implicit declaration of function fax_set_local_identapp_rxfax.c:266: warning: implicit declaration of function fax_set_header_infoapp_rxfax.c:267: warning: implicit declaration of function fax_set_rx_fileapp_rxfax.c:269: warning: implicit declaration of function fax_set_phase_d_handlerapp_rxfax.c:270: warning: implicit declaration of function fax_set_phase_e_handlerapp_rxfax.c:281: warning: implicit declaration of function fax_rx_processapp_rxfax.c:284: warning: implicit declaration of function fax_tx_processapp_rxfax.c:321: warning: passing argument 1 of fax_release from incompatible pointer typemake[1]: *** [app_rxfax.o] Error 1make[1]: Leaving directory `/usr/src/asterisk-1.2.0-beta1/apps'make: *** [subdirs] Error 1 Does the latest asterisk break the fax apps? Any ideas anyone? Thanks, MD ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo Canceling
I just ran the command once. I just called one, and I heard myself in the background, but I did not get that message in dmesg. Still, that message about it being disabled worries me because I get really bad echo on SOME calls. I got that disabled echo cancel message 4 times in my dmesg, but I am not sure when those messages appeared. Tad On 10/4/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 04 October 2005 10:59, Tad Heckaman wrote: zaptel Disabled echo canceller because of tone (rx) on channel 16 I just did dmesg -c and thats what I got... I think there was a call allready in progress. Whats that about? Was that after the first or second dmesg -c? Procedure: dmesg -c place call that will have terrible echo finish call dmesg -c After the 2nd dmesg -c did you see that message? I'm also assuming a relatively unloaded system, and if that on channel 16 was the channel your call went through (You'll see Zap/16-1 or something to that effect in the * CLI) then you've identified the problem. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Tad Heckaman ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] CVS Head - Obtaining Newbie Question
Can someone explain what is meant by CVS Head? Leigh, CVS Head is the latest-and-greatest development version of Asterisk. CVS stands for Concurrent Versioning System and is the system used by the developers to track and coordinate changes in the programming. You can obtain the CVS Head version of Asterisk by following the instructions under the CVS repository section of this page: http://www.asterisk.org/download HOWEVER -- the CVS version is the bleeding, spurting edge of development. It's not recommended for production machines. 1.2 is the beta version and recommended for serious testing and perhaps some non-critical production machines. 1.0.9 is the current, stable, production level release. Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 Subscribe/Notify
BJ Weschke wrote: Upgrade Asterisk. Versions of HEAD post 8-29-05 have this functionality built in. Some of it is currently broken, but there is a patch in the bug tracker that fixes status notification for Eye-beam. haven't tried with Snom. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk w/ BRIstuff compile error
Trying to compile BRIstuff 0.2.0-RC8o. Ran the download.sh and compile.sh scripts to automate the process. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\1.0.9-BRIstuffed-0.2.0-RC8n\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -c -o channel.o channel.c channel.c:64: error: static declaration of 'uniquelock' follows non-static declaration include/asterisk/channel.h:58: error: previous declaration of 'uniquelock' was here --johann ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo Canceling
Well, I dont receive faxes anyway, it goes to a POTS line. I also disabled it in the zconfig.h file and recompiled, but I haven't installed it yet. So going back to my original email... Anything else that might be causing my issues? On 10/4/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 04 October 2005 11:02, Tad Heckaman wrote: After doing a quick search, it appears that maybe a bellsouth echo can is turning my echo can off? How do I tell zaptel to leave it on, regardless if it recieves that tone? The tone to disable echo cancellers is a good thing, not a bad one. Fax machines and modems tend to not work well with echo cancellation (they take care of it themselves) and if you disable the tone detection in zaptel (there's a #define for it in zconfig.h IIRC) then the chances of your faxes/modem calls working through the PRI in either direction will go down significantly. I actually have a wishlist item to enable/disable that tone detection on a per-call basis. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Tad Heckaman ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] FXS static and noise problem
Hi: I have one TDM40B and one TDM04B on my Asterisk box. Both were working fine. Then, all of the FXS ports started to make echo sounds when I make FXS to IAX or SIP connection. All of the FXS ports fail to make bridging to the FXO channels. And when I try to make a call from the console to the FXO ports, I only hear static noise. Any suggestions about the source of the sudden change in behaviour? Regards; Chawki Hammoud __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo Canceling
On Tuesday 04 October 2005 11:26, Matt wrote: You know what.. I have sporadic echo issues too and I just checked my dmesg and also see that! What's this all about? *STOP* You will receive these messages if you send or receive faxes. I asked for this particular procedure to be executed because I was curious to see if zaptel was seeing an echo cancel disable tone when calling the numbers with extreme echo. Again, this is NORMAL to see and EXPECTED if you are sending or receiving faxes. He's not calling a fax machine (I suspect he's not anyway) so I wanted to make sure that the zaptel echocan was NOT hearing the disable tone. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] ADSI -- is it dead? Worth bothering with?
Since Colin Anderson -- in a previous thread -- asked the question about whether ADSI was dead, I thought it was worth discussing. We've been using Nortel Vista 350s in our office up until now. The phones are from Telus, I don't know if there's any way to unlock them. It would appear Telus hasn't done much in the way of updating software for these phones; or if they have, they haven't told us about it. Personally -- the features on this phone have never really worked to my satisfaction, and the only feature I consistently use is the Message Waiting flashing LED. The other ones are just like speed dial functions, and you have to press so many buttons you might as well dial it yourself. Does anybody else have anything to add? -Stephen- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] CVS Head - Obtaining Newbie Question
Thank you! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Pralle Sent: 04 October 2005 11:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CVS Head - Obtaining Newbie Question Can someone explain what is meant by CVS Head? Leigh, CVS Head is the latest-and-greatest development version of Asterisk. CVS stands for Concurrent Versioning System and is the system used by the developers to track and coordinate changes in the programming. You can obtain the CVS Head version of Asterisk by following the instructions under the CVS repository section of this page: http://www.asterisk.org/download HOWEVER -- the CVS version is the bleeding, spurting edge of development. It's not recommended for production machines. 1.2 is the beta version and recommended for serious testing and perhaps some non-critical production machines. 1.0.9 is the current, stable, production level release. Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 1.0.8 and TDM stop acking inbound calls?
Cirelle Enterprises wrote: we also experienced this with asterisk 1.0.9 and rev H of the tdm with 4 fxo modules we were restarting asterisk every night via cron and this still happened in our case, 3 out of 4 fxo modules (2,3,4) crapped out and stopped ack'ing incoming calls (outgoing calls were fine) it took a reboot of the server to get the card operational again and answering calls As a certain Zippy would say: Yow! I assume you reached that point because unloading and reloading the wctdm modules didn't do anything? Do the digital interfaces have these sorts of problems? Is there an alternate FXO solution? I've heard nothing but trouble with the TDM, and I know that's probably because the 99% of satisfied users are generally quiet but still... ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Speed Up SayDigits?
Is there a way to slow down or speed up the speed at which SayDigits rattles off a series of digits? Thanks, Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo Canceling
On Tuesday 04 October 2005 11:54, Tad Heckaman wrote: Well, I dont receive faxes anyway, it goes to a POTS line. I also disabled it in the zconfig.h file and recompiled, but I haven't installed it yet. I thought this was a TE110P, and not a TDM4xx or X101P? So going back to my original email... Anything else that might be causing my issues? If there is no chance of faxing going over this and you are seeing the disabling echo canceller due to tone on channel 'x' whenever you place a call to that exchange, you can disable the tone detector and eliminate the problem. Of course, the best solution is to find out why the telco's sending that signal (or why the other end is, if that's the case). -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] G.729 Codec
Hello, How do I make sure the G.729 codec is being utilized fully and not just as a passthru? I've registered it and followed the install instructions __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Announcing � Voice over IP Directory Services (http://www.voipDS.org)
--- Olle E. Johansson [EMAIL PROTECTED] wrote: Balaji NJL wrote: Announcing Voice over IP Directory Services (http://www.voipDS.org) Sorry to be negative, but this kind of services came up in tons when e-mail was a new service. All of the addresses registred in there is now totally un-usable because of spam. I do not really believe in global directory services as a solution... The main reason for Spamming is because of the fact the way SMTP protocol is designed. It doest verify the sender or it lets receiver to decide who should send email to them. Its not the fault of the Global directory. Thats why in the spec, the receiver, *you*, are in complete control of who shd receive your information. Do you agree that if we address the spamming issue, this would be a viable solution. -B __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] DPH-140S SIP Phone oddities
Hi, list! I'm playing on an [EMAIL PROTECTED] installation, since a month or two. I've had no trouble setting it up 'n running. I've bought a couple of DLINK's DPH-140S SIP Phones, to use with Asterisk. From this phones, I can make receive calls with no trouble, but, when I try to use some interactive function (eg Directory or Voicemail), the phone seems unable to transmit the digits to Asterisk. With the same config, but with a softphone (X-Lite), the digits are transmitted with no trouble at all. Please, do anyone have any clue? Thanks in advance. Juan. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.9/116 - Release Date: 30/09/2005 attachment: winmail.dat___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 phones?
On Tuesday 04 October 2005 06:09, Stephen Bosch wrote: Hi, everyone: I'm in the processing of deciding what IP phones we should use with our Asterisk server, and I wanted to get feedback from the user community on the quality, reliability and ease of operation of Snom phones. What do you have to say about these phones? Are there other phones you'd suggest along with or instead of Snom? Stephen I have used the SNOM190 360, Grandstream budget-tone100, Swissvoice ip10s and EQtel SNOM190 - nice quality, nice sound, expensive, some obscure features which if not turned off cause havoc (Transfer on Hook, Number guessing). Latest firmware solves the annoying loud beep when second call comes in. (SNOM200 is reputedly better...) SNOM360 - total overkill - only for receptionist but nice and expensive. Grandstream - very nice, nice price, has some convenient buttons (Message button dials own extension so easy to set up voicemailmain), attended transfer via flash button Swissvoice ip10s - nice and small, 4 programmable buttons on front so easily customisable, make sure that you have latest firmware (build 12), has 2 lines so no easy way to tell when transferring to phone that the callee is busy on the phone (phone picks up on line 2) EQtel - chunky militaristic look, battery backup (unique), have not used it much, extensive help on connecting to a SIP provider, voice quality ok, enquiring for the settings on the phone are 'spoken' to you rather than displayed on LCD, attended transfer via flash button My current preference is the Grandstream. Regards Paul -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] UPDATE - 512 Calls w/ Dig Rec: NFS Setup and Benchmarks
Tim, Thank you for the information. I will keep it in mind when implementing my mixing and archiving system and share the results with the list when it is complete. Thanks, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer tim panton wrote: On 3 Oct 2005, at 22:54, Matt Roth wrote: List members, It has been a while, but I once implemented a simple shared database over NFS, so dredging my memory produced the following: Future Plans and Unresolved Issues == I wrote Windows software for another project that mixes leg files, indexes them by call time, and archives them after a given period of time. I plan to port that code to a set of shell scripts that will be run on the Digital Recording server out of cron. If anyone knows of an existing project that has accomplished this already, please let me know. Before writing these scripts, I have two questions that need answered: 1) How can I tell when a file is complete on the NFS server? If I recall right, you can't (not on the nfs server end). The way I used to handle this was to have the creating client rename the file once it has finished with it. (remove a leading dot is good). I think you can assume (with a decent NFS implementation) that the rename won't happen untill all the queued writes+close have occurred. 2) What will happen on the NFS client if the NFS server crashes (I expect the leg files to be written to the local mount point until the mount is reesablished)? Nothing so tidy, certainly not on files that were open at the time of the crash. To get that behavior even for new files you would need to un-mount the nfs filesystem on the client whenever there is a crash. (Hmm, kinda like the translucent filesystem...) Tim. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] spandsp and page orientation
Hi Shawn, Could you explain what you mean by 'orientation'. Are your faxes rotated 90 degrees?, are they compressed in the longitudinal plane? Send me one of your landscaped tiff files offlist and I'll try to see whart is going on. Craig - Original Message - From: Shawn Porter [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, October 04, 2005 10:31 PM Subject: [Asterisk-Users] spandsp and page orientation I have just installed spandsp-0.0.2 onto my Asterisk 1.0.9 I am using an old Intel 536EP (actually found drivers that work) BUT...all my faxes are coming in landscape mode Has anyone come across this? any fixes? Shawn ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] G.729 Codec
on asterisk command line do a show translations -bill On 4-Oct-05, at 12:29 PM, Crystal Stream, Incorporated wrote: Hello, How do I make sure the G.729 codec is being utilized fully and not just as a passthru? I've registered it and followed the install instructions __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SPA-3000 generating one-ring calls
On Monday 03 October 2005 13:50, Paul Dugas wrote: This is a wierd one. Can't figure it out. I have an SPA-3000 at the house handling my incoming line. It's setup to direct the incoming call to asterisk. Works great 99% of the time. A few times a day, I'm getting calls that ring once internally and are then hungup. I managed to get a detailed log [1] of what's happening today and it looks to me that the SPA is acting wierd. Can someone verify this for me? I looks to me that the Sipura is just CANCEL'ing the call shortly (2 secs in this example) after setting it up. I'm looking for someone to verify this before I stop looking at Asterisk as the cause and focus on the SPA. Paul I had a simliar problem on a Sipura 1000 unit. It turned out that there was a voicemail message and instead of a stutter tone, the analog phone is rung once. I turned this off by setting VMWI to off. Paul -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] DPH-140S SIP Phone oddities
Title: RE: [Asterisk-Users] DPH-140S SIP Phone oddities Change your DTMF setting to rfc2833. You may be using an incompatible type with your phones. ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Juan Janczuk Sent: Tuesday, October 04, 2005 12:31 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] DPH-140S SIP Phone oddities Hi, list! I'm playing on an [EMAIL PROTECTED] installation, since a month or two. I've had no trouble setting it up 'n running. I've bought a couple of DLINK's DPH-140S SIP Phones, to use with Asterisk. From this phones, I can make receive calls with no trouble, but, when I try to use some interactive function (eg Directory or Voicemail), the phone seems unable to transmit the digits to Asterisk. With the same config, but with a softphone (X-Lite), the digits are transmitted with no trouble at all. Please, do anyone have any clue? Thanks in advance. Juan. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.9/116 - Release Date: 30/09/2005 File: ATT207437.txt This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] G.729 Codec
from the CLI, show g729 -- In my situation, with polycom 501s, if a phone calls another internal phone and canreinvite is set to yes, this does not count against your licenses 'cause the phones are now the only devices in the conversation. you can still find in my phones' status menu what codec it has negotiated for the current call. When asterisk drops from the loop for me, my phones remain on g729. Moj Crystal Stream, Incorporated wrote: Hello, How do I make sure the G.729 codec is being utilized fully and not just as a passthru? I've registered it and followed the install instructions __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo Canceling
Correct, this is a TE110P. I disabled the echocan disable thing in the config file, but I havent actually installed it since it is in production. Is there someway to make a backup of the modules before I reinstall zaptel? I want to easily jump back to the point before I changed some of the settings, incase something becomes messed up. On 10/4/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 04 October 2005 11:54, Tad Heckaman wrote: Well, I dont receive faxes anyway, it goes to a POTS line. I also disabled it in the zconfig.h file and recompiled, but I haven't installed it yet. I thought this was a TE110P, and not a TDM4xx or X101P? So going back to my original email... Anything else that might be causing my issues? If there is no chance of faxing going over this and you are seeing the disabling echo canceller due to tone on channel 'x' whenever you place a call to that exchange, you can disable the tone detector and eliminate the problem. Of course, the best solution is to find out why the telco's sending that signal (or why the other end is, if that's the case). -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Tad Heckaman ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] G.729 Codec
On 10/4/05, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: from the CLI, show g729 -- In my situation, with polycom 501s, if a phone calls another internal phone and canreinvite is set to yes, this does not count against your licenses 'cause the phones are now the only devices in the conversation. you can still find in my phones' status menu what codec it has negotiated for the current call. When asterisk drops from the loop for me, my phones remain on g729. Moj are you saying that when asterisk is used only to connect two phones using g729, then there is no need for a g729 license on asterisk? by the same token, if asterisk was to be used to translated, eg g729 to gsm, then I would need the g729 license. am I understanding it correctly? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] app_rxfax module won't load
I managed to compile app_rxfax and app_txfax against the latest asterisk (1.2 beta 1). When trying to load the app_rxfax module I get this error: [app_rxfax.so]Oct 4 12:52:25 WARNING[3701]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handlerOct 4 12:52:25 WARNING[3701]: loader.c:488 load_modules: Loading module app_rxfax.so failed! or when trying to load app_txfax module I get this error: [app_txfax.so] -- Registered IAX2 to '65.39.205.121', who sees us as 24.103.230.244:4569Oct 4 12:59:16 WARNING[3771]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: fax_set_header_infoas Anybody have a clue what the problem is? By mistake I had spandsp 0.3 installed instead of 0.2 but it has now been changed. I can't find any leftover links etc...HELP! -MD- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
