[asterisk-users] Receive Fax with rxfax on asterisk with debian
Hello, my name is dominik, and i'm using asterisk with voip without isdn, only sip. I'm using Asterisk Version 1.0.7 on Debian 3.0. I've configured the fax receive in the /etc/asterisk/extensions.conf: exten = 99,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif) exten = 99,2,rxfax(${FAXFILE}) exten = 99,3,Hangup In the Debuglevel i see, while i send a fax,that he wants to write the tif file. But on my sending machine i got the error "3311 the number isn't a g3 fax". On asterisk i don't find any errors. When i call the number with a telefon i got the fax sound. Can you help me ? Gruß Dom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MSSQL connection
On 9 Sep 2006, at 00:42, Kevin Smith wrote: Hi everyone, I am looking to log CDR records to our MSSQL database for further examination on the records. From what I gathered from the wiki I have to choose between FreeTDS and unixODBC. Is there a better choice? Which option would be better in the log run? Also configuration asterisk to use both modules. Any good tips on that, I followed the steps provided by the following pages: http://www.voip-info.org/wiki/view/FreeTDS http://www.voip-info.org/wiki/view/Asterisk+app_dbodbc But this is the error I get: (note: some information has been changed for security, such as 'user' and pass was changed to phone) # isql -v MSSQL-astersik phone phone [S1000][unixODBC][FreeTDS][SQL Server]Unable to connect to data source [28000][unixODBC][FreeTDS][SQL Server]Login incorrect. [][unixODBC][FreeTDS][SQL Server]Login failed for user 'phone'. [][unixODBC][FreeTDS][SQL Server]Cannot open database requested in login 'cdr'. Login fails. [ISQL]ERROR: Could not SQLConnect from odbcinst [MSSQL-FreeTDS] Description = FreeTDS ODBC driver for MSSQL Driver = /usr/lib/libtdsodbc.so Setup = /usr/lib/libtdsS.so FileUsage = 1 from odbc [MSSQL-asterisk] description = Asterisk ODBC for MSSQL driver = MSSQL-FreeTDS server = XXX.XXX.XXX.XXX port= 1433 database= cdr user= phone password= phone tds_version = 7.0 language= us_english Maybe I am just overlooking something or there is something that isn't registering with me that is under my nose. Any help would be appreciated. My guess is it is an error between the keyboard and chair ;). We have just been through this - but with Oracle - and came to the conclusion that we didn't want to tightly couple asterisk with the DB, we felt it could be a performance hit - on both sides - plus it meant allowing ODBC traffic over a network we couldn't secure. In the end we got a script written that reads the Master.csv file, turns the new data into XML and does an HTTP Post of the data to a web service running on the Oracle system which parses the XML and inserts the records in the database. We plan to run the script every few minutes (from cron). Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
Mike wrote: Here it is: dialplan dialplan.impossibleMatchHandling=1 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=[7]xx|[9]xxT|[9][1]xxT dialplan.digitmap.timeOut=3/ When I dial 845, I get fast busy. When I dial 9-555-555-, it dials without the need to press send. All good result. Actually, as soon as you hit 8 you will get the fast busy. Is that your full dialplan? What about an emergency (911) or other N11 calls? What about direct dial international calls (011...)? When I dial 9-555-5 and wait, nothing happens So, it looks like what you want is a global dialing timeout in the phone, which the Polycom phones don't appear to have once you break dialtone. But you may be able to kluge the digit timeout to give you that feature if you don't need it for what it is meant for. Right now you are using it to timeout when a digit other than 1 is pressed after the 9. That isn't really necessary (unless 91 followed by 9 digits is actually a valid number for whatever you are doing with it). Also, you are using the brackets unecessarily, since you only have one digit within them. An equivalent dialplan that doesn't use the digit timeout feature would be: digitmap dialplan.digitmap=7xx|9[2-9]x|91xx The digit timeout feature is typically used for direct dial international calls and calling the operator. If you don't need either of those then you could do something like this: digitmap dialplan.digitmap='7xx|9[2-9]x|91xx|[79]x.T dialplan.digitmap.timeOut=15/ which would timeout and send whatever sequence you had pressed after 15 seconds if you hadn't already matched one of the other patterns. Note that asterisk may possibly respond with error code 484 if the sequence pressed isn't complete, which would make the phone continue to ask for more digits. So, the other part of the solution is to add: exten = _X.,1,Congestion() to extensions.conf in the context you are using for your polycom phone(s). That will match anything that doesn't match one of your valid extensions as long as it is two digits or more. So you still will get the behaviour you don't like if someone just presses 7 or 9 and nothing else. But it will give you most of what you want, assuming I understand what you are looking for in the first place (you could try x.T in the digitmap and _. in extensions.conf, but _. is likely to cause other problems). Note: When the Polycom gets the congestion response from Asterisk it plays the congestion tone for only about 3-4 seconds, and then hangs up, which is different behaviour from when you press an 8 for instance. If you want the behaviours to be similar you could do something like this: exten = _X.,1,Answer() exten = _X.,2,Playtones(congestion) exten = _X.,3,Wait(30) exten = _X.,4,Hangup() John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Little help for a newbie configuring a TDM13B - ztcfg fails on channel 4
Hi On Fri, Sep 08, 2006 at 04:54:32PM -0500, Iván Vega R. wrote: Hi everyone, I'm new on Asterisk. One help item: please post messages with a descriptive subject line. Something like: problem configuring zaptel or: ZT_CHANCONFIG failed on channel 4: Invalid argument (22) I'm trying to follow a few tutorials on the net, and fortunately this has been the only stumbling block so far. I do: modprobe zaptel A separate modprobe for zaptel is not needed. The whole point of modprobe is that a module will automatically load modprobe wcfxs modprobe wcfxs modprobe wcfxs modprobe wcfxo You seem to be confused here: You need to load just one kernel module for the TDM400P card: wctdm . wcfxo is for the single-port FXO card X100P. wcfxs is an obsolete version of wctdm. There is no point in loading the same module several times: if you modprobe a module that is already loaded, modprobe will simply do nothing. So why do people run 'modprobe zaptel' separately? One possible and very stupid reason is that they have configured (in /etc/modprobe.conf ) an automatic executionof ztcfg after the load of the module zaptel . It will naturally fail. Just remove any such a line from modprobe.conf . Another less stupid reason is that people run ztcfg automatically after loading the module wctdm. However ztcfg needs to write to some device files that take some time to get generated in later udev-based system. So the fix is to remove the automatic execusion of ztcfg from the post-install action of wctdm as well. I have this on zapata.conf: [channels] busydetect=1 busycount=7 You use Kewlstart signalling. This means that you expect the telco to provide you hangup detection. In such a case: why do you use busydetect? (detecting that a line is busy by hearing busycount busy tones) relaxdtmf=yes relaxdtmf by default? Maybe give the telco some credit? transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 pickupgroup=1-4 The FXO channel is in the same pickup group? immediate=no context=hemac signalling=fxo_s signalling=fxo_ks callerid=asreceived channel=1 channel=2 channel=3 ; shorter method: channel=1-3 group=2 callerid=Batman123 signalling=fxs_ks channel=4 After I run ztcfg -vv I get this: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) 4 channels configured. Changing signalling on channel 4 from Unused to FXO Kewlstart ZT_CHANCONFIG failed on channel 4: Invalid argument (22) Strange this is that if I change the signalling on the first three channels (or the 4th for that matter), the same message appears, so I'm wondering if it's picking up the changes in the configuration... or is there something I'm not getting? The exact modules that were detected on thecard can be shown in the kernel messages that followed the load of the module wctdm . (Or use the script xpp/genzaptelconf to generate a valid zaptel.conf ) -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Forward Problem
I’m currently trying to write a section into my dialplan that when a user dials *78, it will beep 3 times, then wait 10 seconds for the user to enter a 10 digit phone number, then beep 3 more times and put that number into my AsteriskDB. I’m very new to this and I know this is probably very simple but I’m having a very hard time accomplishing this. Here is what I have currently which works for the most part. I thought that using WaitExten(10) would store the numbers that they enter to ${EXTEN} but I was wrong, instead I get just the s. Any idea what I'd have to change here to either store the number to ${EXTEN} or another variable? Any help would be appreciated. [CFWD] ; Call Forward Unconditional exten = *78,1,Set(cfwd_able=${DB(User/${CALLERID(number)}/cfwd_able}) exten = *78,2,GotoIf($[${cfwd_able} = 0]?10) exten = *78,3,GotoIf($[${cfwd_able} = 1]?4) exten = *78,4,Goto(get-fwd,s,1) exten = *78,10,Playback(feature-not-avail-line) exten = *78,11,Hangup exten = *73,1,Set(DB(User/${CALLERID(number)}/callforward)=Disabled) exten = *73,2,DBdel(User/${CALLERID(number)}/callforwardnumber) exten = *73,3,AGI(/etc/asterisk/scripts/cfwdprocess.pl) exten = *73,4,Playtones(!1400/500,!0/250,!1400/500,!0/250,!1400/330) exten = *73,5,Hangup [get-fwd] exten = s,1,NoOp(${TIMESTAMP} get-cfwd begins) exten = s,2,Playtones(!1400/500,!0/250,!1400/500,!0/250,!1400/330) exten = s,2,WaitExten(10) exten = s,3,Set(DB(User/${CALLERID(number)}/callforward)=Enabled) exten = s,4,Set(DB(User/${CALLERID(number)}/callforwardnumber)=${EXTEN}) exten = s,5,AGI(/etc/asterisk/scripts/cfwdprocess.pl) exten = s,6,Playtones(!1400/500,!0/250,!1400/500,!0/250,!1400/330) exten = s,7,Wait(5) exten = s,8,Hangup -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.12.2/442 - Release Date: 9/8/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call forwarding
Hello to all asterisk users, I have a problem with call forwarding. My extensions.conf: [outbound] exten = _*22*XXX,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4}) exten = _*22*,1,DBdel(CFIM/${CALLERID(num)}) Have three stations, 301, 302 and 303. When dial on 301 following number: *22*302 it should redirect all calls targeted to 301 to number 302. But it doesn`t work. If anyone of you has experience with call forwarding, your help will be appreciated. Thank you very much. Vladimir Here is SIP output: --- -- SIP read from 192.168.0.10:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-4b96baec From: 301 sip:[EMAIL PROTECTED];tag=c3d74f8b3bb05e94o0 To: sip:[EMAIL PROTECTED];tag=as3e772365 Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Max-Forwards: 70 Contact: 301 sip:[EMAIL PROTECTED]:5060 User-Agent: Sipura/SPA2002-3.1.2(a) Content-Length: 0 --- (10 headers 0 lines)--- -- SIP read from 192.168.0.10:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-dd489610 From: 301 sip:[EMAIL PROTECTED];tag=c3d74f8b3bb05e94o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username=301,realm=asterisk,nonce=46339822,uri=sip:[EMAIL PROTECTED],algorithm=MD5,response=3284e4149a3abe9e0c4c454af19aa7b5 Contact: 301 sip:[EMAIL PROTECTED]:5060 Expires: 240 User-Agent: Sipura/SPA2002-3.1.2(a) Content-Length: 422 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 437796 437796 IN IP4 192.168.0.10 s=- c=IN IP4 192.168.0.10 t=0 0 m=audio 16406 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (15 headers 19 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.0.10 : 5060 (NAT) Found user '301' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.10:16406 Found description format PCMU Found description format G726-32 Found description format G723 Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format NSE Found description format telephone-event Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x51d (g723|ulaw|alaw| g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for *22*302 in outbound (domain 192.168.0.1) list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (NAT) to 192.168.0.10:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-dd489610;received=192.168.0.10 From: 301 sip:[EMAIL PROTECTED];tag=c3d74f8b3bb05e94o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Executing Set(SIP/301-503d, DB(CFIM/301)=302) in new stack -- Executing Hangup(SIP/301-503d, ) in new stack Reliably Transmitting (NAT) to 192.168.0.10:5060: SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-dd489610;received=192.168.0.10 From: 301 sip:[EMAIL PROTECTED];tag=c3d74f8b3bb05e94o0 To: sip:[EMAIL PROTECTED];tag=as0c8d34d4 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Retransmitting #1 (NAT) to 192.168.0.10:5060: SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-dd489610;received=192.168.0.10 From: 301 sip:[EMAIL PROTECTED];tag=c3d74f8b3bb05e94o0 To: sip:[EMAIL PROTECTED];tag=as0c8d34d4 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.2.9 compile error
Same problem here on CentOS 4.4 :( Strange that apparently the tarball was not tested if it would even compile On Fri, 8 Sep 2006, Stuart Sheldon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I can confirm the same problem, it looks like the oct612x directory tree is missing from the tarball Stu Bill Maidment wrote: Hi I've just tried to compile the zaptel-1.2.9 release and I get the following error: HOSTCC /usr/local/src/zaptel-1.2.9/wct4xxp/fw2h /usr/local/src/zaptel-1.2.9/wct4xxp/fw2h /usr/local/src/zaptel-1.2.9/wct4xxp/OCT6114-128D.ima /usr/local/src/zaptel-1.2.9/wct4xxp/vpm450m_fw.h make[3]: *** No rule to make target `/usr/local/src/zaptel-1.2.9/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h', needed by `/usr/local/src/zaptel-1.2.9/wct4xxp/vpm450m.o'. Stop. make[2]: *** [/usr/local/src/zaptel-1.2.9/wct4xxp] Error 2 make[1]: *** [_module_/usr/local/src/zaptel-1.2.9] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.17-1.2174_FC5-i686' make: *** [linux26] Error 2 zaptel-1.2.8 compiled OK. So what has changed? Did I do something wrong? Cheers Bill - -- Randomly Generated Fortune Tag: Many pages make a thick book. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) iD8DBQFFAlUkK69Y+xPZrWYRApE/AJ947hGNhPOnHVVojbg/8X2kvPTKgQCgn0+d N9FEkMIUKqol8Lru+N+ByxE= =Gr3f -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Outgoing Spool Failed with ViciDial (MattF?)
lol Matt Riddell (IT) wrote: Arun Kumar wrote: hi thanks for reply. I'm using vicidial to make calls at 2.0 dial level it is able to make calls but when I see the asterisk -r most of the time it shows Outgoing Spool Failed. Which Spool File ? Er, probably the best place to ask would be the VICIdial forum on mattf's website, unless he wants to chime in? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cheers Andrea Andrea Cristofanini Gedam Europe Srl Gedam Advanced Communication Ltd Torino, Italy C.so Re Umberto 21 Mobile : + 39 329 1871756 PSTN: + 39 011 19824516 FreeVoip: 6838601 http://www.gedameurope.com http://freevoip.gedameurope.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.2.9 compile error
Remco Barendse wrote: Same problem here on CentOS 4.4 :( Strange that apparently the tarball was not tested if it would even compile I just tried to apply the 1.2.9 patch to 1.2.8 and that fails to patch. Looks like someone had a bad day -- Bill Maidment Maidment Enterprises Pty Ltd www.maidment.com.au si hoc non legere potes tu asinus es ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [asterisk-users] How to Install H323
hello , Which channel do you want to set chan_h323 chan_oh323 or chan_ooh323 ? Harry --- Wasif [EMAIL PROTECTED] a écrit : Hello, Could anyone tell me how to install/configure H323 with Asterisk 1.2.11 . Thanks Wazb ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Outgoing Spool Failed with ViciDial (MattF?)
http://www.eflo.net/VICIDIALforum and VICIDIAL does not use call files for Originate spooling. It uses the Manager API. MATT--- On 9/8/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Arun Kumar wrote: hi thanks for reply. I'm using vicidial to make calls at 2.0 dial level it is able to make calls but when I see the asterisk -r most of the time it shows Outgoing Spool Failed. Which Spool File ? Er, probably the best place to ask would be the VICIdial forum on mattf's website, unless he wants to chime in? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFAdhzS6d5vy0jeVcRAnPFAJ0SOVbMh+nwaaFf4NYzB2F9dNAHSACfRi9b uPYjG7ZiNRKU5uw9hfT0qs0= =DnnW -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Forwarding in SIP.conf
Thanks all. It works fine now. -- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---BeginMessage--- Check your Dial() string to make sure that you haven't mistyped and put gafachi-o instead of gafachi-out. Specifiying the full host name will also work. As a hint, you can refresh these changes with out restarting your server (and therefore without disrupting any calls in progress) extensions reload will refresh the extensions file reload will reload all your configs sip reload will reload only sip configs (and re-register everything) Very handy when working on an active machine. On September 8, 2006 14:19, [EMAIL PROTECTED] wrote: Tim, this is the way I have Gafachi set up in sip.conf and works well with channels that have an ATA attached to it but not the virtual one. I have changed the host in extensions.conf to the .sip.gafachi.com. But I have calls on the server and cannot restart it yet. I'll keep you posted and thanks for the feedback. [gafachi-out] type=peer secret=xx username=x fromuser=x fromdomain=xxx host=.sip.gafachi.com ;usereqphone=yes; This provider requires ;user=phone on URI ;nat=yes rtptimeout=60 dtmfmode=rfc2833 -- Original message -- From: Tim St. Pierre [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpybFLtTio6d.pgp Description: PGP signature ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Processing Slow 11 seconds
I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What don't I get about SIP?
Actually, as soon as you hit 8 you will get the fast busy. Is that your full dialplan? What about an emergency (911) or other N11 calls? What about direct dial international calls (011...)? Its my full test dialplan for now. I do get fast busy as soon as I hit 8, so that part works. So, it looks like what you want is a global dialing timeout in the phone, which the Polycom phones don't appear to have once you break dialtone. But you may be able to kluge the digit timeout to give you that feature if you don't need it for what it is meant for. Right now you are using it to timeout when a digit other than 1 is pressed after the 9. That isn't really necessary (unless 91 followed by 9 digits is actually a valid number for whatever you are doing with it). Also, you are using the brackets unecessarily, since you only have one digit within them. An equivalent dialplan that doesn't use the digit timeout feature would be: digitmap dialplan.digitmap=7xx|9[2-9]x|91xx Fair enough, but that doesn't solve the original issue, but it makes my kludge a bit better [lots of good info removed] Note that asterisk may possibly respond with error code 484 if the sequence pressed isn't complete, which would make the phone continue to ask for more digits. So, the other part of the solution is to add: exten = _X.,1,Congestion() That will match anything that doesn't match one of your valid extensions as long as it is two digits or more. So you still will get the behaviour you don't like if someone just presses 7 or 9 and nothing else. But it will give you most of what you want, assuming I understand what you are looking for in the first place (you could try x.T in the digitmap and _. in extensions.conf, but _. is likely to cause other problems). Did I misread the Asterisk wiki pages, because I believed that when a pattern was present, the pattern takes precedence over any real extensions? (i.e. if I have both 1234 and _1XXX as extensions in a context)? Thanks John, I appreciate all the info. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Another (quick) Polycom 501 question
Hi all, That's my last one for a while (I hope). How can I (if at all possible) make the 501 turn on the speaker phone as soon as a digit is dialed (if the handset is not lifted)?Sort of likewhat a normal speakerphone does. The reason I want this is I want the 501 digitmap to be taken into consideration even if the handset isnt lifted and the speakerphone button isn't consciously pressed. For all those users who don't want to press send, but like dialing without lifting the handset (and can't be bothered to press the speakerphone button). Yes I know it's capricious, but we have the users we have... Yes, I have read the admin manual, but couldn't find the info. I am assuming I just don't know what to look for, but that this functionality exists. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stupid question about FXS/FXO
Hi Ivan. As you see in this page: http://www.neobits.com/do/dtls?pid=9583 This card is a bundle, wich means supports both, FXO, and FXS. FXO ports should be used to connect your 3 telco lines, and FXS port to connect some phones. Regards On 9/8/06, Iván Vega R. [EMAIL PROTECTED] wrote: Hi yet again, Is a TDM13B card the correct one if I have three phone lines and I want to use the extra port to connect a normal phone? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stupid question about FXS/FXO
Yeah I figured after some experimentation :) Thanks. I got confused while configuring zaptel and asterisk, hehe. On 9/9/06, Moises Silva [EMAIL PROTECTED] wrote: Hi Ivan. As you see in this page: http://www.neobits.com/do/dtls?pid=9583 This card is a bundle, wich means supports both, FXO, and FXS. FXO ports should be used to connect your 3 telco lines, and FXS port to connect some phones. Regards On 9/8/06, Iván Vega R. [EMAIL PROTECTED] wrote: Hi yet again, Is a TDM13B card the correct one if I have three phone lines and I want to use the extra port to connect a normal phone? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN / Multiplink PPP (ZapRAS)
Hello, I have found and read Steven Critchfields writeup on how to use ZapRas (Thanks Critch!), however I am a bit confused. His write up is here: http://copilotconsulting.com/mail-archives/asterisk.2003/msg01030.html Currently we have a full PRI (23B Channels, 1D) coming into our Asterisk Box using a TE110P. Voice calls work great, and I have it all configured up. Now, I would also like to be able to use this box to establish a 128k (possibly 256k multilink) PPP connection to an ISP who supports 128k (2B's) or 256k (4B's) connection. One thing I don't get understand is, is the Asterisk/Linux box the modem? I am confused when he states that he used a Ascend Pipeline 75 as I don't really understand how that fits into the equation. Can't asterisk dial our end ISP, and then I can set up routing at the linux level to interface with ppp0 device for example? Any help would be greatly appreciated! Thanks so much, - Brent ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DID Provider in Thailand
Does anyone know of a DID provider in Thailand? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MSSQL connection
Thanks Tim, That was my first thought as well but then I thought, might as well give it a try. But it is turning into a hassle more then anything. I already have a PHP script wrote to for MySQL so the conversion to MSSQL shouldn't be bad. Thanks, Kevin Tim Panton wrote: On 9 Sep 2006, at 00:42, Kevin Smith wrote: Hi everyone, I am looking to log CDR records to our MSSQL database for further examination on the records. From what I gathered from the wiki I have to choose between FreeTDS and unixODBC. Is there a better choice? Which option would be better in the log run? Also configuration asterisk to use both modules. Any good tips on that, I followed the steps provided by the following pages: http://www.voip-info.org/wiki/view/FreeTDS http://www.voip-info.org/wiki/view/Asterisk+app_dbodbc But this is the error I get: (note: some information has been changed for security, such as 'user' and pass was changed to phone) # isql -v MSSQL-astersik phone phone [S1000][unixODBC][FreeTDS][SQL Server]Unable to connect to data source [28000][unixODBC][FreeTDS][SQL Server]Login incorrect. [][unixODBC][FreeTDS][SQL Server]Login failed for user 'phone'. [][unixODBC][FreeTDS][SQL Server]Cannot open database requested in login 'cdr'. Login fails. [ISQL]ERROR: Could not SQLConnect from odbcinst [MSSQL-FreeTDS] Description = FreeTDS ODBC driver for MSSQL Driver = /usr/lib/libtdsodbc.so Setup = /usr/lib/libtdsS.so FileUsage = 1 from odbc [MSSQL-asterisk] description = Asterisk ODBC for MSSQL driver = MSSQL-FreeTDS server = XXX.XXX.XXX.XXX port= 1433 database= cdr user= phone password= phone tds_version = 7.0 language= us_english Maybe I am just overlooking something or there is something that isn't registering with me that is under my nose. Any help would be appreciated. My guess is it is an error between the keyboard and chair ;). We have just been through this - but with Oracle - and came to the conclusion that we didn't want to tightly couple asterisk with the DB, we felt it could be a performance hit - on both sides - plus it meant allowing ODBC traffic over a network we couldn't secure. In the end we got a script written that reads the Master.csv file, turns the new data into XML and does an HTTP Post of the data to a web service running on the Oracle system which parses the XML and inserts the records in the database. We plan to run the script every few minutes (from cron). Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Forwarding in SIP.conf
I have a follow up question. How do I pass on the caller ID of the call I'm forwarding to the other party? I can pass on the channels caller ID but prefer to pass on the forwarding party's number instead. -- Original message -- From: [EMAIL PROTECTED] Thanks all. It works fine now. -- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---BeginMessage--- ---BeginMessage--- Check your Dial() string to make sure that you haven't mistyped and put gafachi-o instead of gafachi-out. Specifiying the full host name will also work. As a hint, you can refresh these changes with out restarting your server (and therefore without disrupting any calls in progress) extensions reload will refresh the extensions file reload will reload all your configs sip reload will reload only sip configs (and re-register everything) Very handy when working on an active machine. On September 8, 2006 14:19, [EMAIL PROTECTED] wrote: Tim, this is the way I have Gafachi set up in sip.conf and works well with channels that have an ATA attached to it but not the virtual one. I have changed the host in extensions.conf to the .sip.gafachi.com. But I have calls on the server and cannot restart it yet. I'll keep you posted and thanks for the feedback. [gafachi-out] type=peer secret=xx username=x fromuser=x fromdomain=xxx host=.sip.gafachi.com ;usereqphone=yes; This provider requires ;user=phone on URI ;nat=yes rtptimeout=60 dtmfmode=rfc2833 -- Original message -- From: Tim St. Pierre [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpz1rsmZopyp.pgp Description: PGP signature ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell Poweredge SC430 and Digium cards compatability enquiry
Hello Matthew, It depends on the chipset on the mainboard. I had problems with a SC1420, the only way to solve it was to get a new server (without Intel chipset). So don't try a chipset which is listed on the Digium compatibility site. Wednesday, September 6, 2006, 8:55:58 AM, you wrote: We're looking at using a number of Dell Poweredge SC430 servers as Asterisk hosts in our smaller overseas offices with Digium cards in to provide local breakout over the pre-existing analogue or digital phone lines (One office uses ISDN2 the others analogue) I note that the SC420 is listed as incompatible but the SC430 appears to be a slightly different beast in terms of chipset, the 430 has the newer E7230 as opposed to the E7221 - does this make a difference to compatibility? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Forwarding in SIP.conf
If you don't set the callerID in the channel, it will get passed on as-is. Don't change it, and it will stay the same. -TIm On September 9, 2006 12:27, [EMAIL PROTECTED] wrote: I have a follow up question. How do I pass on the caller ID of the call I'm forwarding to the other party? I can pass on the channels caller ID but prefer to pass on the forwarding party's number instead. -- Original message -- From: [EMAIL PROTECTED] Thanks all. It works fine now. -- Original message -- From: Tim St. Pierre [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpt5tN7gJk5i.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Intel Based G.729 and SVN-trunk-r42453
Hi,I was testing the intel based G729 codec on SVN-trunk-r42453 following the new instructions for compiling with SVN trunk and it in preliminary tests it works ok for some calls but I found when one end of the call is an IVR or Music On Hold the sound gets all distorted and asterisk segfaults. You can view the backtrace at http://pastebin.ca/165220Any assistance on this would be appreciated.-- Regards,Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Forwarding in SIP.conf
I tried both of them but it still goes asID unavailable. First I commented it out, that did not work and left it blank and that did not work either. Below is the sample in sip.conf [4305]type=frienduser=4305secret=xxx;context=from-sipcallerid= ; left it blank but did not get passed on!host=dynamicnat=yesqualify=yescanreinvite=nodtmfmode=rfc2833 -- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---BeginMessage--- If you don't set the callerID in the channel, it will get passed on as-is. Don't change it, and it will stay the same. -TIm On September 9, 2006 12:27, [EMAIL PROTECTED] wrote: I have a follow up question. How do I pass on the caller ID of the call I'm forwarding to the other party? I can pass on the channels caller ID but prefer to pass on the forwarding party's number instead. -- Original message -- From: [EMAIL PROTECTED] Thanks all. It works fine now. -- Original message -- From: Tim St. Pierre [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpxqRvOtLBQ4.pgp Description: PGP signature ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call Processing Slow 11 seconds
In case you use an adapter or voip phone: Did you try to press hash # after the number ? - thenthe adapter/voip phonedials immediately and doesnt wait for the next digit timeout. Cheers Gerry -Original MessageFrom: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of [EMAIL PROTECTED]Sent: Samstag, 9. September 2006 15:15To: asterisk-users@lists.digium.comSubject: [asterisk-users] Call Processing Slow 11 seconds I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453
Jason Lee wrote: Hi, I was testing the intel based G729 codec on SVN-trunk-r42453 following the new instructions for compiling with SVN trunk and it in preliminary tests it works ok for some calls but I found when one end of the call is an IVR or Music On Hold the sound gets all distorted and asterisk segfaults. You can view the backtrace at http://pastebin.ca/165220 Any assistance on this would be appreciated. Have you compiled with debugging symbols instead of CPU optimization? Can you type `bt' after the segfault, to give us some more detail? How long into the call does this happen? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call Processing Slow 11 seconds
Yes that works. I'm using Linksys adapter, is there a code I can put in the dial plan to prevent users from putting # after the number? I have a lot of people on the server and cannot ask them all to be pushing # after every call. Thanks for the tip and any help will be appreciated. -- Original message -- From: "G.Jacobsen" [EMAIL PROTECTED] In case you use an adapter or voip phone: Did you try to press hash # after the number ? - thenthe adapter/voip phonedials immediately and doesnt wait for the next digit timeout. Cheers Gerry -Original MessageFrom: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of [EMAIL PROTECTED]Sent: Samstag, 9. September 2006 15:15To: asterisk-users@lists.digium.comSubject: [asterisk-users] Call Processing Slow 11 seconds I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3 ---BeginMessage--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Processing Slow 11 seconds
Yes you could script a dialplan putting ... and S0 (zero) at the end. An example : (xxS0) It will dial 6 digits directly when you enter the 6th. You could learn how to adapt your Linksys dialplan looking this wiki. http://voip.wikispaces.com/ [EMAIL PROTECTED] escribió: Yes that works. I'm using Linksys adapter, is there a code I can put in the dial plan to prevent users from putting # after the number? I have a lot of people on the server and cannot ask them all to be pushing # after every call. Thanks for the tip and any help will be appreciated. -- Original message -- From: G.Jacobsen [EMAIL PROTECTED] In case you use an adapter or voip phone: Did you try to press hash # after the number ? - then the adapter/voip phone dials immediately and doesnt wait for the next digit timeout. Cheers Gerry -Original Message *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] *Sent:* Samstag, 9. September 2006 15:15 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Call Processing Slow 11 seconds I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3 Asunto: RE: [asterisk-users] Call Processing Slow 11 seconds De: G.Jacobsen [EMAIL PROTECTED] Fecha: Sat, 9 Sep 2006 17:20:05 + Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453
I recompiled with debuging options...both bt and btfull outputs http://pastebin.ca/165250Before I recompiled it gave me a second of audio then I got nothing but distortion for 5 seconds then asterisk would crash. I retested after compiling it with just a call between two local devices one using ulaw and the other using g729 and I'm getting nothing but distortion. I then tried calling music on hold and it took 3 minutes to crash the whole time I got nothing but distortion. On 9/9/06, Daniel Pocock [EMAIL PROTECTED] wrote: Jason Lee wrote: Hi, I was testing the intel based G729 codec on SVN-trunk-r42453 following the new instructions for compiling with SVN trunk and it in preliminary tests it works ok for some calls but I found when one end of the call is an IVR or Music On Hold the sound gets all distorted and asterisk segfaults. You can view the backtrace at http://pastebin.ca/165220 Any assistance on this would be appreciated.Have you compiled with debugging symbols instead of CPU optimization?Can you type `bt' after the segfault, to give us some more detail? How long into the call does this happen?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards,Jason LeeOmegaServ [EMAIL PROTECTED]Direct Line: (204) 480-1238Toll Free: (866) 664-7786 Ext 200http://www.omegaserv.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
Mike wrote: Did I misread the Asterisk wiki pages, because I believed that when a pattern was present, the pattern takes precedence over any real extensions? (i.e. if I have both 1234 and _1XXX as extensions in a context)? It's the opposite. Asterisk always uses the most specific match for an extension, i.e. anything that matches _1XXX will take precedence over _, but if it matches _12XX that will take precedence over _1XXX, etc. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453
Jason Lee wrote: I recompiled with debuging options... both bt and btfull outputs http://pastebin.ca/165250 Before I recompiled it gave me a second of audio then I got nothing but distortion for 5 seconds then asterisk would crash. I retested after compiling it with just a call between two local devices one using ulaw and the other using g729 and I'm getting nothing but distortion. I then tried calling music on hold and it took 3 minutes to crash the whole time I got nothing but distortion. This suggests that someone/something gave the command `stop now' Can you send the backtrace from a segfault? On 9/9/06, Daniel Pocock [EMAIL PROTECTED] wrote: Jason Lee wrote: Hi, I was testing the intel based G729 codec on SVN-trunk-r42453 following the new instructions for compiling with SVN trunk and it in preliminary tests it works ok for some calls but I found when one end of the call is an IVR or Music On Hold the sound gets all distorted and asterisk segfaults. You can view the backtrace at http://pastebin.ca/165220 Any assistance on this would be appreciated. Have you compiled with debugging symbols instead of CPU optimization? Can you type `bt' after the segfault, to give us some more detail? How long into the call does this happen? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453
Sorry about that. I thought I had the right core dump. I retried again and the output from bt and bt full is at http://pastebin.ca/165289It took 1min 50seconds of nothing but distortion before asterisk segfaulted -- Regards,JasonOn 9/9/06, Daniel Pocock [EMAIL PROTECTED] wrote: Jason Lee wrote: I recompiled with debuging options... both bt and btfull outputs http://pastebin.ca/165250 Before I recompiled it gave me a second of audio then I got nothing but distortion for 5 seconds then asterisk would crash. I retested after compiling it with just a call between two local devices one using ulaw and the other using g729 and I'm getting nothing but distortion. I then tried calling music on hold and it took 3 minutes to crash the whole time I got nothing but distortion.This suggests that someone/something gave the command `stop now'Can you send the backtrace from a segfault? On 9/9/06, Daniel Pocock [EMAIL PROTECTED] wrote: Jason Lee wrote: Hi, I was testing the intel based G729 codec on SVN-trunk-r42453 following the new instructions for compiling with SVN trunk and it in preliminary tests it works ok for some calls but I found when one end of the call is an IVR or Music On Hold the sound gets all distorted and asterisk segfaults. You can view the backtrace at http://pastebin.ca/165220 Any assistance on this would be appreciated. Have you compiled with debugging symbols instead of CPU optimization? Can you type `bt' after the segfault, to give us some more detail? How long into the call does this happen? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What don't I get about SIP?
It certainly makes sense, and I tried it...it works, you are right. So what do you make of this page : http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf +sorting Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Marvin Sent: September 9, 2006 2:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: Did I misread the Asterisk wiki pages, because I believed that when a pattern was present, the pattern takes precedence over any real extensions? (i.e. if I have both 1234 and _1XXX as extensions in a context)? It's the opposite. Asterisk always uses the most specific match for an extension, i.e. anything that matches _1XXX will take precedence over _, but if it matches _12XX that will take precedence over _1XXX, etc. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Another (quick) Polycom 501 question
Hi Mike, As far as I know, you need to at least start the dialing (ie New call, speaker, etc) for the digitmap to even come into play. The only settings that I am aware of that you can try to change are dialplan.impossibleMatch-Handling and dialplan.digitmap from sip.conf. Kevin Mike wrote: Hi all, That's my last one for a while (I hope). How can I (if at all possible) make the 501 turn on the speaker phone as soon as a digit is dialed (if the handset is not lifted)? Sort of like what a normal speakerphone does. The reason I want this is I want the 501 digitmap to be taken into consideration even if the handset isnt lifted and the speakerphone button isn't consciously pressed. For all those users who don't want to press send, but like dialing without lifting the handset (and can't be bothered to press the speakerphone button). Yes I know it's capricious, but we have the users we have... Yes, I have read the admin manual, but couldn't find the info. I am assuming I just don't know what to look for, but that this functionality exists. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 54
hi i need helpl configuring a quintum tenor analog gateway using sip with asterisk. anyone, help is appreciated the model of the gteway is asm200 i need the settings to configure it with asterisk. for some reason it registers with asterisk but when try to call the extension from the quintum it is not recognized. help help help thanks From: [EMAIL PROTECTED] Reply-To: asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 26, Issue 54 Date: Sat, 9 Sep 2006 12:00:25 -0700 (MST) MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc2-f18.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Sat, 9 Sep 2006 12:03:59 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 3C2CA41D5;Sat, 9 Sep 2006 12:00:25 -0700 (MST) X-Message-Info: LsUYwwHHNt1Qrly5/IdcOLxnJ5Hdz4bhYGyQtYHi6jU= X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 09 Sep 2006 19:04:00.0431 (UTC) FILETIME=[B57B13F0:01C6D442] Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: Call Forwarding in SIP.conf ([EMAIL PROTECTED]) 2. RE: Call Processing Slow 11 seconds (G.Jacobsen) 3. Re: Intel Based G.729 and SVN-trunk-r42453 (Daniel Pocock) 4. RE: Call Processing Slow 11 seconds ([EMAIL PROTECTED]) 5. Re: Call Processing Slow 11 seconds (Alberto Sagredo) 6. Re: Intel Based G.729 and SVN-trunk-r42453 (Jason Lee) 7. Re: What don't I get about SIP? (John Marvin) 8. Re: Intel Based G.729 and SVN-trunk-r42453 (Daniel Pocock) 9. Re: Intel Based G.729 and SVN-trunk-r42453 (Jason Lee) 10. RE: What don't I get about SIP? (Mike) -- Message: 1 Date: Sat, 09 Sep 2006 17:12:54 + From: [EMAIL PROTECTED] Subject: Re: [asterisk-users] Call Forwarding in SIP.conf To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Skipped content of type multipart/alternative-- next part -- An embedded message was scrubbed... From: Tim St. Pierre [EMAIL PROTECTED] Subject: Re: [asterisk-users] Call Forwarding in SIP.conf Date: Sat, 9 Sep 2006 16:52:40 + Size: 2109 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/828bebdd/attachment-0001.eml -- Message: 2 Date: Sat, 9 Sep 2006 19:17:23 +0300 From: G.Jacobsen [EMAIL PROTECTED] Subject: RE: [asterisk-users] Call Processing Slow 11 seconds To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii In case you use an adapter or voip phone: Did you try to press hash # after the number ? - then the adapter/voip phone dials immediately and doesnt wait for the next digit timeout. Cheers Gerry -Original Message From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Samstag, 9. September 2006 15:15 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call Processing Slow 11 seconds I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3 -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/632afcb4/attachment-0001.htm -- Message: 3 Date: Sat, 09 Sep 2006 18:23:37 +0100 From
Re: [asterisk-users] Call Processing Slow 11 seconds
Thanks, I tried that and did not work for me. My users are calling US number and without the # at the end of the last digit dials it takes 11 seconds before it starts ringing. -- Original message -- From: Alberto Sagredo [EMAIL PROTECTED] Yes you could script a dialplan putting ... and S0 (zero) at the end. An example : (xxS0) It will dial 6 digits directly when you enter the 6th. You could learn how to adapt your Linksys dialplan looking this wiki. http://voip.wikispaces.com/ [EMAIL PROTECTED] escribió: Yes that works. I'm using Linksys adapter, is there a code I can put in the dial plan to prevent users from putting # after the number? I have a lot of people on the server and cannot ask them all to be pushing # after every call. Thanks for the tip and any help will be appreciated. -- O riginal message -- From: "G.Jacobsen" <[EMAIL PROTECTED]> In case you use an adapter or voip phone: Did you try to press hash # after the number ? - then the adapter/voip phone dials immediately and doesnt wait for the next digit timeout. Cheers Gerry -Original Message *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] *Sent:* Samstag, 9. September 2006 15:15 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Call Processing Slow 11 seconds I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been conside ring u sing openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3 Asunto: RE: [asterisk-users] Call Processing Slow 11 seconds De: "G.Jacobsen" <[EMAIL PROTECTED]> Fecha: Sat, 9 Sep 2006 17:20:05 + Para: "Asterisk Users Mailing List - Non-Commercial Discussion"Para: "Asterisk Users Mailing List - Non-Commercial Discussion" ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing li st To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.2.9 compile error
--- Bill Maidment [EMAIL PROTECTED] wrote: Hi I've just tried to compile the zaptel-1.2.9 release and I get the following error: Same here, using CentOS 4.4 kernel 2.6.9-42.0.2.ELsmp, got these errors when compiling zap: make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found make[3]: *** No rule to make target `/usr/src/zaptel/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h', needed by `/usr/src/zaptel/wct4xxp/vpm450m.o'. Stop. make[2]: *** [/usr/src/zaptel/wct4xxp] Error 2 make[1]: *** [_module_/usr/src/zaptel] Error 2 make: *** [linux26] Error 2 Hope someone has a workaround for this problem __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems configuring Polycom 301
HiI have successfully been running with several Polycom SoundPoint 501phones and recently purchased some Polycom 301 phones.However, I can't seem to get the phones to register. The phone seesthe asterisk server, but all calls our are busy. The only difference for 'sip show peer xxx' for a working 501 phone anda non working 301 phone is:asterisk1*CLIAddr-IP : 192.168.80.204 Port 5060 # 501Addr-IP : (Unspecified) Port 0 # 301 'sip show peers' returns:asterisk1*CLI sip show peers Name/usernameHostDyn Nat ACL Port Status720/720(Unspecified)D0UNKNOWN 712/712192.168.8.205 D5060 OK (80 ms) 711/711192.168.8.203 D5060 OK (84 ms) 710/710192.168.8.204 D5060 OK (98 ms)Any 301 configuration tips would be appreciated.Thanks-- Jim Freeze ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.2.9 compile error
The work around is at: http://www.sineapps.com/news.php?rssid=1496 On 09/09/06, Samy Antoun [EMAIL PROTECTED] wrote: --- Bill Maidment [EMAIL PROTECTED] wrote: Hi I've just tried to compile the zaptel-1.2.9 release and I get the following error: Same here, using CentOS 4.4 kernel 2.6.9-42.0.2.ELsmp, got these errors when compiling zap: make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found make[3]: *** No rule to make target `/usr/src/zaptel/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h', needed by `/usr/src/zaptel/wct4xxp/vpm450m.o'. Stop. make[2]: *** [/usr/src/zaptel/wct4xxp] Error 2 make[1]: *** [_module_/usr/src/zaptel] Error 2 make: *** [linux26] Error 2 Hope someone has a workaround for this problem ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztdummy installed but choppy audio warning on load
On a new set up Centos 4.4, kernel 2.6.9-42.0.2.EL, yum updated, 2 BRI-HFC cards, no digium hardware. modprobe zaptel and modprobe ztdummy are both in rc.local, and lsmod gives: [EMAIL PROTECTED] ~]# lsmod Module Size Used by ztdummy 3924 0 zaptel206852 5 ztdummy When asterisk starts it logs warnings: Sep 9 20:28:00 WARNING[2645] res_musiconhold.c: Unable to open pseudo channel for timing... Sound may be choppy. Sep 9 20:28:02 WARNING[2645] chan_iax2.c: Unable to open IAX timing interface: No such file or directory I've Googled the error message, but to no avail. Any thoughts, please? nigel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy installed but choppy audio warning on load
zap show status will tell you if Asterisk is really using ztdummy Make sure you have chan_zap.so enabled in modules.conf (or that it isn't disabled with a noload declaration) Nigel Godfrey wrote: On a new set up Centos 4.4, kernel 2.6.9-42.0.2.EL, yum updated, 2 BRI-HFC cards, no digium hardware. modprobe zaptel and modprobe ztdummy are both in rc.local, and lsmod gives: [EMAIL PROTECTED] ~]# lsmod Module Size Used by ztdummy 3924 0 zaptel206852 5 ztdummy When asterisk starts it logs warnings: Sep 9 20:28:00 WARNING[2645] res_musiconhold.c: Unable to open pseudo channel for timing... Sound may be choppy. Sep 9 20:28:02 WARNING[2645] chan_iax2.c: Unable to open IAX timing interface: No such file or directory I've Googled the error message, but to no avail. Any thoughts, please? nigel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.2.9 compile error
--- Nigel Godfrey [EMAIL PROTECTED] wrote: The work around is at: http://www.sineapps.com/news.php?rssid=1496 Thanks, I'll give it a try. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Scope of contexts
Hi all, I am trying to understand contexts a bit better. The problem I have is when you know when a context is finished. Is this when a new context starts? Example: [context1] exten = _9170X,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED]) [context2] exten = 6394,1,Dial(Local/6275/n) When your call starts at context1, will it automatically go to context2 when context1 is finished? Hope someone can shed some light on this Thanks, Rene ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
Mike wrote: It certainly makes sense, and I tried it...it works, you are right. So what do you make of this page : http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf +sorting Interesting. I got my information from Asterisk: The Future of Telephony (in the dialplan chapter). Perhaps the wiki page refers to 1.0 behaviour, and 1.2 behaviour is what is defined in Asterisk: TFOT? My experimentation so far has shown the Asterisk: TFOT information to be correct. I haven't played around with #include, which the wiki says can change the dialplan extension sorting. I may have to experiment with that to see if it has any effect. I hope that what I said is correct regardless, because it makes the most sense and is less likely to cause weird issues when changing the order of #includes, etc. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scope of contexts
Rene wrote: Hi all, I am trying to understand contexts a bit better. The problem I have is This should help: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/c650.html Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scope of contexts
Yep, that should help, and the short answer to you question is NO. Regards On 9/9/06, Doug Lytle [EMAIL PROTECTED] wrote: Rene wrote: Hi all, I am trying to understand contexts a bit better. The problem I have is This should help: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/c650.html Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GX-2000, doesn't send calls to free lines
Thanks Daniel, your advice helped. It was the call waiting not working, which I thoughtwas working because I had selected 'No' beside 'Disable Call-Waiting'. But for call waiting to work properly, I also needed to select 'No' beside ' Enable Call Features' and then dial *70 to enable it on Asterisk.Now it works perfect. On 9/8/06, Daniel Salama [EMAIL PROTECTED] wrote: You need to enable call waiting on the phone's config. - Daniel On Sep 8, 2006, at 9:35 AM, Zeeshan Zakaria wrote: First call is answered by LINE1, but if this line is still busy and a second call comes in, it doesn't go to LINE2, instead called listens asterisk message, all lines are busy, please leave your message after the tone. I tried resetting phone to factory default setting too, but still it does the same. Same extension if configured on X-TEN, it works with no problems for all available free lines. Grandstream phone should go upto 11 lines and only for 12th call should say that lines are busy. What I need to configure in this Grandstream phone which I haven't figured out yet. I've firmware 1.1.0.16 -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to send correct Caller ID on PRI
I figured out how to send the caller ID, it is working. But now I am trying to send the Company Name along with caller ID, and that is not working. What could be the reason for that. I am using SerCallerID. Is there something else I should use to send the alphabets? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Grandstream GX-2000 phones for paging
I am still trying to make it work. Where did you get firmware version 1.1.1.9. On there website they have only 1.1.0.16 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream, how to use the configuration tool
I'll try to use it, but in future I think I'll get some other phones with better configuration utilities. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Roundrobin not working on PRI
I sent the incoming calls to different queues, and now everything is working fine. Just changed the MoH option of the first queue to ring tone, so that the callers hears the ring going, otherwise they'll feel uncomfortable why music has started all of a sudden. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using option 'r' in queue doesn't announce frequeny etc.
In my queues, I wanted callers to listen dial tone going, instead of listening the music, to I used option 'r' in queue command. Now it doesn't announce the position of caller and estimated hold time etc. Is this normal for this setting or I am soing something wrong. If I don't use option r and leave queue as it is, i.e. playing MoH, then all announcements work ok.-- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Whcih phones are better for mass deployment
I am having hard time with grandstream phones for a30 phone setup. When a change in configuration is required, I have to change their configurations manually for almost all of them. Their configuration utility is not very straight forward to use. For my next installation, I would prefer some other phones with better configuration and remote accress utility. My question to those of you with more experience, what IP phones are better for mass deployment and easy management of updates and configurations? Or what other solution is better for mass deployment of phones? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using option 'r' in queue doesn't announce frequeny etc.
On 18:47, Sat 09 Sep 06, Zeeshan Zakaria wrote: In my queues, I wanted callers to listen dial tone going, instead of listening the music, to I used option 'r' in queue command. Now it doesn't announce the position of caller and estimated hold time etc. Is this normal for this setting or I am soing something wrong. If I don't use option r and leave queue as it is, i.e. playing MoH, then all announcements work ok. I noticed the same thing couple of months ago. Talked about it here and on irc but noone could give me a clear answer. I'm now still stuck with moh because I really want the announcements to be played :( If someone has an mp3 or wav of ringing, please post it... -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Whcih phones are better for mass deployment
Try the Linksys ATA. I gave up on Granstream and have 4 sitting in around. -- Original message -- From: "Zeeshan Zakaria" [EMAIL PROTECTED] I am having hard time with grandstream phones for a30 phone setup. When a change in configuration is required, I have to change their configurations manually for almost all of them. Their configuration utility is not very straight forward to use. For my next installation, I would prefer some other phones with better configuration and remote accress utility. My question to those of you with more experience, what IP phones are better for mass deployment and easy management of updates and configurations? Or what other solution is better for mass deployment of phones? -- Zeeshan A Zakaria ---BeginMessage--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Processing Slow 11 seconds
[EMAIL PROTECTED] wrote: Thanks, I tried that and did not work for me. My users are calling US number and without the # at the end of the last digit dials it takes 11 seconds before it starts ringing. If you are dialing 11 digits then set the Linksys Dial Plan to: (1xx), the phone will dial out right after the last digit. You will need to modify the dial plan if there are other patterns as well that are expected. -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to send correct Caller ID on PRI
Zeeshan Zakaria wrote: I figured out how to send the caller ID, it is working. But now I am trying to send the Company Name along with caller ID, and that is not working. What could be the reason for that. I am using SerCallerID. Is there something else I should use to send the Most phone companies will not allow for you to set callerid name, only number. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom new firmware and bootrom
Before the phone starts it boot process, I edit the phone settings under server information, enter the IP of the sip server and also the protocol to use, UDP Only. Chris David Gagnon wrote: Hi Chris, I'm would like to get more information about this problem. Why the phone isn't registering properly with the new firmware and what did you mean by hard coding the sip settings? Thx David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Chris Dos Envoyé : 7 septembre 2006 15:30 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Polycom new firmware and bootrom Not to mention the feature that the new firmware and bootrom that prevent it from registering with the Asterisk server unless you hard code the sip settings. Chris Jessee J Holmes wrote: Also keep in mind that as of right now, the latest bootrom and firmware available from Polycom (and thus your reseller) are Bootrom 3.2.2 and Firmware 2.0.1 The 2.0.1 firmware is new as of a day or two and include some enhancements for buddy lists and shared presence as well as newly added secured TLS support (according to Polycom). Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 7, 2006, at 1:19 PM, Douglas Garstang wrote: That process is worse than pulling teeth! -Original Message- *From:* Jessee J Holmes [mailto:[EMAIL PROTECTED] *Sent:* Thursday, September 07, 2006 11:25 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Polycom new firmware and bootrom All authorized Polycom resellers will have access to this firmware and are required to provide this firmware to you. Contact the reseller you purchased the Polycom phone from. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 7, 2006, at 11:55 AM, Bruce Reeves wrote: Typically you have to go to a reseller who you purchased Polycom equipment from. Even then it can be tricky since they have to find away to get you the files with out upsetting Polycom. On 9/7/06, *Douglas Garstang* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Polycom are analy retentive about giving out software updates. -Original Message- From: Nathan Alberti [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] Sent: Thursday, September 07, 2006 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom new firmware and bootrom Stupid question where did you find it ? Looked at their site downloads and under the extranet site but could only see old versions. Nathan. On 07/09/2006, at 10:21 AM, Chris Dos wrote: Well, it seems that Polycom has release new firmware 2.0.1 and bootrom 3.2.2. I've proceded to upgrade all my ip430 phones because they were essentially broken with the original firmware. All the phones boot up fine now, grab their files. They just won't talk to the asterisk server any more. I just figured out that I need to hard code the sip server and tell it to talk udp only. After this, the phones worked again. Any idea on what I need to configure to fix the phones so they will know which server to talk to and only talk to it via udp? Chris ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth
Re: [asterisk-users] Whcih phones are better for mass deployment
Polycom & Aastra are both great in this manner. Michael --Original Message Text--- From: [EMAIL PROTECTED] Date: Sun, 10 Sep 2006 00:18:37 + Try the Linksys ATA. I gave up on Granstream and have 4 sitting in around. -- Original message -- From: "Zeeshan Zakaria" [EMAIL PROTECTED] I am having hard time with grandstream phones for a 30 phone setup. When a change in configuration is required, I have to change their configurations manually for almost all of them. Their configuration utility is not very straight forward to use. For my next installation, I would prefer some other phones with better configuration and remote accress utility. My question to those of you with more experience, what IP phones are better for mass deployment and easy management of updates and configurations? Or what other solution is better for mass deployment of phones? -- Zeeshan A Zakaria --NextPart_Webmail_9m3u9jl4l_7096_1157847517_1-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What really happens between Asterisk and an SPA-3000?
I'm trying to get a clear understanding just how calls are routed in a mixed SPA3k and Asterisk system. This is my present (incomplete) understand and I'd appreciate any corrections. I'm especially interested in what happens between Asterisk and an SPA3k. Note: - POTSaudio refers to POTS line audio signal. SIPaudio refers to sip IP packets. User is us with the Sipura and Asterisk goodies. Caller is the outside person calling us. No power to SPA3k: -- failover mode. FXO line (CO line) audio is connected directly to FXS line (POTS phones) by a relay. Incoming SIP call to Asterisk: -- Handled by Asterisk as SIP to SIP. Call never touches SPA3k. Incoming POTS call to SPA3k: POTSaudio converted to SIPaudio signal in SPA3k. SIPaudio forwarded to PSTN-IN extension on Asterisk server (ext 201 for me). Asterisk should ring both SIP phones and POTS phones using context in extensions.conf so User can pickup either. If POTS phone picked up: Does POTSaudio go directly back and forth over POTS line or is there a SIP conversion anywhere? If SIP phone picked up: --- SIPaudio from SPA3k Caller is heard by User SIPaudio from User goes out over PSTN-OUT to SPA3k which converts SIPaudio to POTSaudio and out Line to Caller. Outgoing calls: --- If 7 digit local or 911, outgoing context in extensions.conf routes call to SPA3k and out PSTN-SPA3k gateway. SIPaudio to SPA3k which converts it to POTSaudio. Other calls are routed either to SIP extensions or SIP provider. SPA3k is out of the picture. Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Whcih phones are better for mass deployment
Can you explain a little bit what make them better for mass deployment. Do they have Windows based software to communicate with all the installed phones and upgrade them and also to remotely monitor them. Is there a separate cost for these software tools or are they free? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Grandstream GX-2000 phones for paging
I finally made the paging to work. But the only thing which I had to change was the number to dial. As in the instructions, it is _**1, but it didn't work for me and I used 333 instead. All other settings are the same. Now the reception phone's one Speed Dial key is assigned ext 333, pressing which activates speaker phones of all the phones in the office for paging. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.2.9 compile error
Nigel Godfrey wrote: The work around is at: http://www.sineapps.com/news.php?rssid=1496 Thank you. I'd forgotten about SVN. Works like a charm. Cheers Bill -- Bill Maidment Maidment Enterprises Pty Ltd www.maidment.com.au si hoc non legere potes tu asinus es ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Whcih phones are better for mass deployment
for mass deployment the Linksys will allow you to update your routers with a tftp server.. You can have the routers always download their software from the tftp server, that way you have the latest on the server for upgrade software. The reason that I don't like granstream is their bad customer support, they live you on your own basically. -- Original message -- From: "Zeeshan Zakaria" [EMAIL PROTECTED] Can you explain a little bit what make them better for mass deployment. Do they have Windows based software to communicate with all the installed phones and upgrade them and also to remotely monitor them. Is there a separate cost for these software tools or are they free? ---BeginMessage--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scope of contexts
Nope. A context ends when a new one starts. The only way for a call to continue is to have a maching extension, and the next higher priority. If you want a call to continue in another context, you need to use the Goto() application. On September 9, 2006 19:04, Rene wrote: Hi all, I am trying to understand contexts a bit better. The problem I have is when you know when a context is finished. Is this when a new context starts? Example: [context1] exten = _9170X,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED]) [context2] exten = 6394,1,Dial(Local/6275/n) When your call starts at context1, will it automatically go to context2 when context1 is finished? Hope someone can shed some light on this Thanks, Rene ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgp2N6znO9LaO.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Whcih phones are better for mass deployment
I am a really big fan of Aastra phones. It's a splinter company of Northern Telecom, so their quality is very good. Provisioning is done via a text file on either a tftp or an ftp server. There is a global file and a per phone file. When you have a good set of config files built, you can include an option for the phones to check the files every day at a predetermined time and reboot if there are any changes. You can also send an SIP NOTIFY to cause the phones to update their config if you change something and need it applied immediately. There is no config utility needed, as the files are human readable. There is an encryption utility if you are concerned about security. When you deploy a new phone, you need only set the TFTP server address. After that, the phone can get all it's settings from the server. I have about 40 of them deployed at client sites that I usually don't have access to. I can change everything from here. Sound quality is great, most of them support PoE and have a passthru ethernet port. The displays are backlit, there is a full duplex speaker phone and headset jack on all models. There is also a built in directory function that loads from a .csv file on the server. BLF support is good on the 9133i and the 480i. I can't say enough good things about these phones. Manufacturer support is also very good. Free firmware downloads from the website and good documentation. -Tim On September 9, 2006 18:51, Zeeshan Zakaria wrote: I am having hard time with grandstream phones for a 30 phone setup. When a change in configuration is required, I have to change their configurations manually for almost all of them. Their configuration utility is not very straight forward to use. For my next installation, I would prefer some other phones with better configuration and remote accress utility. My question to those of you with more experience, what IP phones are better for mass deployment and easy management of updates and configurations? Or what other solution is better for mass deployment of phones? -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpUZT8t5RNms.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Whcih phones are better for mass deployment
I'll speak on the Aastra, since that is what I know, although most of this applies to Polycom as well. There is no windows software needed at all. Personally, I haven't been a Microsoft customer in more than half a decade. Their operating systems are not appropriate for telecom applications. You can do all your configuration with a text editor. This is good for several reasons: 1) You can administer the phone config directly on the server over an ssh connection. 2) A shell script can create phone config files. I have a shell script that appends to extensions.conf, sip.conf, voicemail.conf, and creates a phone config file. You can automate things very easily this way. 3) Since there is only one setting to put into the phone to use the remote config, a customer can be talked through a factory reset and reset the server address over the phone if they really screw things up. You can have an inventory of phones with your config server address already set. All you need is the phone MAC address, and you can build a config file. This means that you could send phones to customers without them having decided what to do with them yet. 4) These phones are reliable and well constructed. They will require less maintenance, and will last longer. 5) They have features that are appropriate to a business environment. 6) They will usually find their way around a NAT firewall, so they are essential plug-and-play at the customer site. Let me know if you have any more questions. -Tim On September 9, 2006 23:01, Zeeshan Zakaria wrote: Can you explain a little bit what make them better for mass deployment. Do they have Windows based software to communicate with all the installed phones and upgrade them and also to remotely monitor them. Is there a separate cost for these software tools or are they free? -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpKBfHPOMykJ.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Whcih phones are better for mass deployment
Thanks for the info. In my next installation, I'll try those phones. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Whcih phones are better for mass deployment
On Saturday 09 September 2006 18:51, Zeeshan Zakaria wrote: For my next installation, I would prefer some other phones with better configuration and remote accress utility. My question to those of you with more experience, what IP phones are better for mass deployment and easy management of updates and configurations? Or what other solution is better for mass deployment of phones? Polycom. Nothing I've ever seen or used works better for mass deployment. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GX-2000 Remote Login Problem
On my grandstream phones, I am having problem that I can't login in remotely. With in the local network, they work ok, but outside the network, they don't. All the NAT settings are ok and I can log into the Asterisk server from X-Ten, no problem. But Grandstream phone doesn't login. I've port forwardedSIP 5060-5080, RTP 1-14000. In rtp.conf, ports range from 1-14000. Same extensions when configured on x-ten, with RTP listening on 1, they login with no problem. But when configured on grandstream, they don't configure. What am I missing here.-- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GX-2000 Remote Login Problem
I've noticed that Grandstream works better using stun and not port forwarding your router. Try setting stun.xten.com or stun.fwdnet.net in your GS2000 and make sure sip.conf has nat=yes. It should work fine. Also, i've noticed that Linksys wireless with speed booster has something in it that is blocking VoIP. I set DMZ, done port forwarding... nothing... I could get it to register but no return sound... it was being blocked. Just something to keep in the back of your mind. bp On 9/10/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: On my grandstream phones, I am having problem that I can't login in remotely. With in the local network, they work ok, but outside the network, they don't. All the NAT settings are ok and I can log into the Asterisk server from X-Ten, no problem. But Grandstream phone doesn't login. I've port forwardedSIP 5060-5080, RTP 1-14000. In rtp.conf, ports range from 1-14000. Same extensions when configured on x-ten, with RTP listening on 1, they login with no problem. But when configured on grandstream, they don't configure. What am I missing here.-- Zeeshan A Zakaria ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Streaming audio for MoH
I am trying some streaming radio for the frist time, but it is not working. I have something like this in my musiconhold.conf: stream = quietmp3:/var/lib/asterisk/mohmp3/stream, http://www.streamaudio.com/listen/default_gonew.asp?headertext=Owner=Empire%20Broadcastingstation=BEET_IRPortalFormat=OptIn=streamtype=filename=Owner=BEET_IR I also tried this stream = quietmp3:/var/lib/asterisk/mohmp3/stream,http://www.shoutcast.com/sbin/shoutcast-playlist.pls?rn=3281file=filename.pls I can see MoH starting, and then it stops in about 15 sec, and I don't listen anything. I dial extension 466 to activate MoH, using exten = 466,1,Answerexten = 466,n,SetMusicOnHold,streamexten = 466,n,WaitMusicOnHold,300exten = 466,n,Hangup It works fine for class default, but the streaming link in class stream doesn't work. Is this link not correct or I am doing something wrong. The file stream.mp3 does exist in mohmp3/stream folder. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to send correct Caller ID on PRI
Actualy it doesn't realy have to do with the phone company not letting you set it, although sometimes between 2 PRIs what you set as the name might come up. It has to do with the fact that when phone companies send callerid to a ringing phone they don't look at what you set for name, they look up the name for the phone number you set using a CNAM query. On 9/9/06, Doug Lytle [EMAIL PROTECTED] wrote: Zeeshan Zakaria wrote: I figured out how to send the caller ID, it is working. But now I am trying to send the Company Name along with caller ID, and that is not working. What could be the reason for that. I am using SerCallerID. Is there something else I should use to send the Most phone companies will not allow for you to set callerid name, only number. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Quintum tenor configuration with asterisk help
[127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 3C2CA41D5;Sat, 9 Sep 2006 12:00:25 -0700 (MST) X-Message-Info: LsUYwwHHNt1Qrly5/IdcOLxnJ5Hdz4bhYGyQtYHi6jU= X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:asterisk-us [EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:asterisk-us [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 09 Sep 2006 19:04:00.0431 (UTC) FILETIME=[B57B13F0:01C6D442] Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: Call Forwarding in SIP.conf ([EMAIL PROTECTED]) 2. RE: Call Processing Slow 11 seconds (G.Jacobsen) 3. Re: Intel Based G.729 and SVN-trunk-r42453 (Daniel Pocock) 4. RE: Call Processing Slow 11 seconds ([EMAIL PROTECTED]) 5. Re: Call Processing Slow 11 seconds (Alberto Sagredo) 6. Re: Intel Based G.729 and SVN-trunk-r42453 (Jason Lee) 7. Re: What don't I get about SIP? (John Marvin) 8. Re: Intel Based G.729 and SVN-trunk-r42453 (Daniel Pocock) 9. Re: Intel Based G.729 and SVN-trunk-r42453 (Jason Lee) 10. RE: What don't I get about SIP? (Mike) -- Message: 1 Date: Sat, 09 Sep 2006 17:12:54 + From: [EMAIL PROTECTED] Subject: Re: [asterisk-users] Call Forwarding in SIP.conf To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] net Content-Type: text/plain; charset=us-ascii Skipped content of type multipart/alternative-- next part -- An embedded message was scrubbed... From: Tim St. Pierre [EMAIL PROTECTED] Subject: Re: [asterisk-users] Call Forwarding in SIP.conf Date: Sat, 9 Sep 2006 16:52:40 + Size: 2109 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/828bebd d/attachment-0001.eml -- Message: 2 Date: Sat, 9 Sep 2006 19:17:23 +0300 From: G.Jacobsen [EMAIL PROTECTED] Subject: RE: [asterisk-users] Call Processing Slow 11 seconds To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii In case you use an adapter or voip phone: Did you try to press hash # after the number ? - then the adapter/voip phone dials immediately and doesnt wait for the next digit timeout. Cheers Gerry -Original Message From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Samstag, 9. September 2006 15:15 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call Processing Slow 11 seconds I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3 -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060909/632afcb 4/attachment-0001.htm -- Message: 3 Date: Sat, 09 Sep 2006 18:23:37 +0100 From: Daniel Pocock [EMAIL PROTECTED] Subject: Re: [asterisk-users] Intel Based G.729 and SVN-trunk-r42453 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Jason Lee wrote: Hi, I was testing the intel based G729 codec on SVN-trunk-r42453 following the new instructions for compiling with SVN trunk and it in preliminary tests it works ok for some calls but I found when one end of the call is an IVR or Music On Hold the sound gets all distorted and asterisk segfaults. You can view the backtrace at http://pastebin.ca/165220 Any assistance