RE: [Asterisk-Users] RH9 and h323.conf
OK. I tried with 1.12.2 and indeed problem fixed. Thanks, Tjapko.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: Sbado, 13 de Diciembre de 2003 04:54 a.m.
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RH9 and h323.conf
SW wrote:
This did not work for me, so I tried
exten = 91x,1,Dial(h323/h323:${Exten:[EMAIL PROTECTED]) and it
worked.
If you upgrade to Open H.323 v1.12.2 this problem is fixed.
Jeremy McNamara
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Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
On Fri, 2003-12-12 at 09:29, Dan wrote: Hi, - Original Message - From: Alastair Maw [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, December 12, 2003 4:58 PM Subject: Re: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-) On 12/12/03 13:56, Dan wrote: This is because the fax is transmitted using the audio stream. It is not related to the signaling protocol (SIP/IAX etc.) but to the audio codec used. Fax uses FSK modulation to transmit the data. If you compress this in a lossy way (GSM, MP3, whatever) then the integrity of the data is affected (more or less seriously depending on the codec used). Fax machines are generally quite picky, so compressing faxes is unlikely to work. I'm wondering why on earth you want to push fax data over a VoIP link at all. Fax compression isn't very efficient. Who wants that??? By fax data I mean the data contained in a fax (basically a picture file), not the fax data audio stream. It can be converted (GIF or JPG) then sent reliable over a slow IP link. Just a special codec at both ends, able to pass the data to the fax app or a fax machine connected to a TDM400/ AT or whatever. It would be much less bandwidth intensive to decode the fax and send it over as proper data rather than audio, compressed using gzip/gif/png/something else. This is exactly what would be great to have it. Sounds like you want app_rcfax at the pstn to decode the fax data, then it would somehow transfer the image file to your local machine via the slow link and use app_tcfax to send the image to your local fax machine. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voice mail - sip:notify message
Hi folks, To provide MWI, * will send out a sip:notify message to the UA. The originator of this message is asterisk, as shown below; NOTIFY sip:[EMAIL PROTECTED]:5065 SIP/2.0 Via: SIP/2.0/UDP 66.121.xxx.yyy:5060;branch=z9hG4bK0466cb21 From: asterisk sip:[EMAIL PROTECTED];tag=as0ffb1bdc === To: sip:[EMAIL PROTECTED]:5065 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 38 Messages-Waiting: yes Voicemail: 4/11 Is there a way that I can change this 'originator' to a numeric value ? Cheers SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Garbled VoiceMail
I tried again at runlevel 3 but to no avail. I'm pretty sure I have sufficient horsepower since I'm running on a box with half gig memory and a speedy CPU. burak I run on a Pentium I /100 Mhz, 32MB RAM with RedHat 9.0 and have no trouble with voicemail audio or Music On Hold. This is a total SIP/IAX(2) machine with no interfaces to the PSTN. Granted this is a much smaller machine than reccomended but it does work. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Party in Paris
SATURDAY 20th I have had far fewer emails than the noise created earlier about Mark's arrival in Paris. Everyone who has contacted me I have replied to once. Again please - if you want to come please email me at [EMAIL PROTECTED] I will set a venue and time tomorrow evening to email to all concerned. Steve - Original Message - From: marrandy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, December 11, 2003 7:58 PM Subject: Re: [Asterisk-Users] * Party in Paris On Thursday 11 December 2003 12:53 pm, Bob Knight wrote: Is the party at the Paris Hilton? sorry, couldn't help it... -- Bob Knight Bob...I'm really surprised at you !!! I thought you would have said, 'Is the party in Paris Hilton' lol ;-) -- Q: What's hard going in and soft and sticky coming out? A: Chewing gum. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] new CVS Checkout
Today I deleted the files in the asterisk, libpri, zaptel directories that are in /usr/src and did a new CVS checkout (not update). After doing the make installs and starting asterisk the show version is the same as before: Asterisk CVS-10/09/03-20:33:57 built by [EMAIL PROTECTED] on a i586 running Linux It looks like there are executable asterisk files in /usr/sbin and /usr/lib with a change date of today. I would have expected a newer data on the Version. Is there something I missed doing? Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new CVS Checkout
Hi, From: rnc Info Lists [EMAIL PROTECTED] Today I deleted the files in the asterisk, libpri, zaptel directories that are in /usr/src and did a new CVS checkout (not update). After doing the make installs and starting asterisk the show version is the same as before: Asterisk CVS-10/09/03-20:33:57 built by [EMAIL PROTECTED] on a i586 running Linux You must be sure you deleted the file .version (hidden) in the .../src/asterisk directory. Better delete all, including directories. See my problem from a couple of days ago in the mail archive. If you have just deleted the content of the asterisk, zapata, zaptel and libpri directories then is normal to have the same version displayed when starting Asterisk. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
On Sat, 2003-12-13 at 05:14, Dan wrote: Hi, - Original Message - From: Steven Critchfield [EMAIL PROTECTED] Sounds like you want app_rcfax at the pstn to decode the fax data, then it would somehow transfer the image file to your local machine via the slow link and use app_tcfax to send the image to your local fax machine. Yup! This is what would be great to exist. I want to be able to reliable transfer faxes between Asterisk servers using slow links, but still keeping the standard fax functionality locally. While I haven't tested it much, my first use of app_rxfax and app_txfax worked fine. The glue is little more than procmail and sample.call combined with working mail servers. I have faith that Steve's work will continue to make the functionality better and it will work more often with the variety of fax machines out there. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] new CVS Checkout
I just updated yesterday, but I did a complete rm -Rf for all of the following directories: /usr/src/zaptel /usr/src/zapata /usr/src/libpri /usr/src/asterisk Then I did a new cvs checkout for all four of those items before recompiling them (make clean; make install) in the same order. My Asterisk now states that its running Version CVS-12/12/03-09:47:51. Hope this helps. Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Sent: Saturday, December 13, 2003 8:31 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] new CVS Checkout Hi, From: rnc Info Lists [EMAIL PROTECTED] Today I deleted the files in the asterisk, libpri, zaptel directories that are in /usr/src and did a new CVS checkout (not update). After doing the make installs and starting asterisk the show version is the same as before: Asterisk CVS-10/09/03-20:33:57 built by [EMAIL PROTECTED] on a i586 running Linux You must be sure you deleted the file .version (hidden) in the .../src/asterisk directory. Better delete all, including directories. See my problem from a couple of days ago in the mail archive. If you have just deleted the content of the asterisk, zapata, zaptel and libpri directories then is normal to have the same version displayed when starting Asterisk. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.550 / Virus Database: 342 - Release Date: 12/9/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.550 / Virus Database: 342 - Release Date: 12/9/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exit the Directory Application?
On Friday 12 December 2003 22:04, Ulexus wrote: On Thursday, 13 November, 2003 11:34, Tilghman Lesher wrote: On Thursday 13 November 2003 07:31, Marcus Adolfsson wrote: How does a user exit the directory application? Say he can't find the person that he is looking for and wants to return the main menu, how would I configure 0 to act this way? Just enter a new extension. For example, if you want # to exit the Directory application, program the # extension. exten = #,1,Goto(s,5) The directory is generated from the voicemail.conf, so I imagine you would also have to an entry for extension '#' to voicemail.conf as well. Don't imagine. Try it. This seems like a really cheap (if effective and expedient) way of doing it. Just a note (and I really should add this to bugs.digium.com, I suppose), both the Directory and the Voicemail2 apps have very myopic view of the rest of the dial-plan or even their current context. Namely, the lack of an escape condition for the Directory and lack of most any dial-out conditions (i.e., '0' or another extension number) in Voicemail2. Directory does not need an escape condition. If you fail to enter anything within the allotted time (see ResponseTimeout), you jump to the t extension. In a production environment, it is far better to take them as a proof-of-concept/development base and customize them to your overall setup than to use them out of the box. We use Voicemail() out of the box in multiple production environments. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new CVS Checkout
Did you do a make clean first before the make make install ? Just my $.02 :) -bh Quoting rnc Info Lists [EMAIL PROTECTED]: Today I deleted the files in the asterisk, libpri, zaptel directories that are in /usr/src and did a new CVS checkout (not update). After doing the make installs and starting asterisk the show version is the same as before: Asterisk CVS-10/09/03-20:33:57 built by [EMAIL PROTECTED] on a i586 running Linux It looks like there are executable asterisk files in /usr/sbin and /usr/lib with a change date of today. I would have expected a newer data on the Version. Is there something I missed doing? Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] new CVS Checkout
On Sat, 2003-12-13 at 16:41, Joe Dennick wrote: I just updated yesterday, but I did a complete rm -Rf for all of the following directories: /usr/src/zaptel /usr/src/zapata /usr/src/libpri /usr/src/asterisk Then I did a new cvs checkout for all four of those items before recompiling them (make clean; make install) in the same order. My Asterisk now states that its running Version CVS-12/12/03-09:47:51. I use make update. In the Makefile under update: is the instruction rm -f .version -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
Hi, From: Steven Critchfield [EMAIL PROTECTED] While I haven't tested it much, my first use of app_rxfax and app_txfax worked fine. The glue is little more than procmail and sample.call combined with working mail servers. I have faith that Steve's work will continue to make the functionality better and it will work more often with the variety of fax machines out there. I have done tests with several fax machines. txfax works great will all of them, including software, but rxfax works for me with just a single old Samsung fax machine. Anyway, it is a great application and the integration with asterisk is good enough for receiving faxes, but to send them is a little bit cumbersome Steve, if you read this, keep up the good work and thanks. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] new CVS Checkout
On Sat, 2003-12-13 at 16:41, Joe Dennick wrote: I just updated yesterday, but I did a complete rm -Rf for all of the following directories: /usr/src/zaptel /usr/src/zapata /usr/src/libpri /usr/src/asterisk Then I did a new cvs checkout for all four of those items before recompiling them (make clean; make install) in the same order. My Asterisk now states that its running Version CVS-12/12/03-09:47:51. I use make update. In the Makefile under update: is the instruction rm -f .version -- Dave Cotton [EMAIL PROTECTED] Thanks to everyone who replied to this question. I got rid of the hidden files and reloaded from CVS. It is ok now.Guess over the next days I'll go through the archives and learn a bit more about using CVS to maintain updates. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Garbled VoiceMail
Burak, Try connecting to your * server with a SIP phone like X-Lite or an IAX phone like DIAX. Do you get the same results with those phones too? Jonathan Burak Balasaygun wrote: Hi, I just got started with asterisk and am having a problem with voice quality. When connecting via either a GS IP phone or calling from the PSTN (via 100XP FXO) the demo-congrats msg I get is garbled (but discernable). Any ideas? thanx burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mysql CDR
Hi! The line with ;sock=/tmp/mysql.sock, i think you must write it without the ;. You need this socket to connect with mysql. Best regards, Mireia -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of listas iPfone Sent: sexta-feira, 12 de dezembro de 2003 16:47 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Mysql CDR Hi all I just installed the mysql cdr support and my database is not registering the calls :( using show modules i see that the cdr_csv.so and the cdr_addon_mysql.so are loaded It is necessary to unload the cdr_csv.so? how to do it? in crd_mysql.conf i have: [global] hostname=localhost dbname=asteriskcdrdb password=new_password user=asteriskcdruser ;port=3306 ;sock=/tmp/mysql.sock i copied the crd_mysql.conf to the /etc/asterisk directory..it is to be there ..or not? and in modules.conf i have: load = cdr_addon_mysql.so It is correct? something more is needed? ( i created the database and table from wiki instructions) How can i know if asterisk is or not trying to register the calls to the database? Thanks! miklos iPFONE Telefonia IP Rua Caio Graco 735 São Paulo SP iPBX +55 11 3801-3702 UK +44 870 - 3403539 FWD 64662 sip:[EMAIL PROTECTED] www.ipfone.com.br [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mysql CDR
On Saturday 13 December 2003 11:02, Mireia Munoz de jesus wrote: The line with ;sock=/tmp/mysql.sock, i think you must write it without the ;. You need this socket to connect with mysql. You don't usually need that configuration line. It's only there if your server and client conflict about the correct location for the sock file. For example, the MySQL default location is /tmp/mysql.sock, but RedHat (and others) insist on putting that in /var/lib/mysql/mysql.sock. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SPA-2000 is shipping, discount for asterisk-users
Some people on this group may have understood from messages posted here that the Sipura SPA-2000 is not currently available for shipping. That is not the case. Voxilla.com has the Sipura SPA-2000 available for immediate shipping, and has had them since late November. The price is $109.95, and it comes with a month of free VoicePulse service with activation fees waived (a $65 value). In return for the tremendous help that the asterisk-users board provides to the community (and because were huge fans of asterisk and Digium), starting now and through Dec. 20th, Voxilla will offer a $10 discount to asterisk-users list subscribers. That makes the price for the SPA-2000 with a month of VoicePulse service to $99.95, plus shipping. There is no handling fee and typical ground UPS shipping rates within the U.S. for a unit range between $6.00 and $8.00. If interested, visit the Voxilla Store at http://voxilla.com/shop. When purchasing a unit, enter *users (without the quote marks) in the coupon code field, and a $10 discount will be applied. Marcelo Rodriguez Editor/Publisher Voxilla.com
Re: [Asterisk-Users] Garbled VoiceMail
Jonathan, I have it sorted it out. Rebooted the box and it works fine on all interfaces now. thanx burak On Sat, 13 Dec 2003 08:40:26 -0800, Jonathan Tew wrote Burak, Try connecting to your * server with a SIP phone like X-Lite or an IAX phone like DIAX. Do you get the same results with those phones too? Jonathan Burak Balasaygun wrote: Hi, I just got started with asterisk and am having a problem with voice quality. When connecting via either a GS IP phone or calling from the PSTN (via 100XP FXO) the demo-congrats msg I get is garbled (but discernable). Any ideas? thanx burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users rgds burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wrong voicemail after transfer?
I'm using a modified default config file for extensions.conf, the one that uses macro-stdexten to handle the stations. We use a TDM30 card for our stations. When a call that has been rung in using that macro transfers the call things work just fine as far as the other instrument ringing. But once the ring timeout has expired, the call then drops into the *original station's* voicemail. E.g. Tammy picks up the call and hookflashes, then dials Jim's extension. Jim's phone rings for 20 seconds. But if he's not at his desk, the call then goes to *Tammy's* voicemail. I've gone through the WIKI and mail archives looking for the solution, but it's sort of hard to conjure the correct search string, I have found. Hopefully I've paid my newbie dues thereby, and someone more clueful might be willing to help. . . Thanks. I have CLI traces if that would be necessary for someone to see. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wrong voicemail after transfer?
Their is an open bug on this on bugs.digium.com it doesn't happen on # transfers just flash transfers from what I can tell. bkw On Sat, 13 Dec 2003, Brian Capouch wrote: I'm using a modified default config file for extensions.conf, the one that uses macro-stdexten to handle the stations. We use a TDM30 card for our stations. When a call that has been rung in using that macro transfers the call things work just fine as far as the other instrument ringing. But once the ring timeout has expired, the call then drops into the *original station's* voicemail. E.g. Tammy picks up the call and hookflashes, then dials Jim's extension. Jim's phone rings for 20 seconds. But if he's not at his desk, the call then goes to *Tammy's* voicemail. I've gone through the WIKI and mail archives looking for the solution, but it's sort of hard to conjure the correct search string, I have found. Hopefully I've paid my newbie dues thereby, and someone more clueful might be willing to help. . . Thanks. I have CLI traces if that would be necessary for someone to see. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New User Questions
Timothy Costello wrote: and somewhere (maybe on the wiki) should be a link to ESR's How to Ask Smart Questions: http://www.catb.org/~esr/faqs/smart-questions.html I know it's been posted to the list several times. It should be part of the FAQ to read it before asking questions... Added link on the FAQ http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wrong voicemail after transfer?
Hi! When a call that has been rung in using that macro transfers the call things work just fine as far as the other instrument ringing. But once the ring timeout has expired, the call then drops into the *original station's* voicemail. This is a known bug: http://bugs.digium.com/bug_view_page.php?bug_id=617 Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
Tilghman Lesher wrote: On Friday 12 December 2003 07:25, Dan wrote: Hi, It would be great if the IAX protocol will be able to tranfer fax data (even converted in another format) between Asterisk boxes, using low bandwidth codecs like GSM. I know that this is possible only with the G.711 now (passing faxes using the audio stream), but maybe in the future...some native support will permit this. You're not going to get that working because GSM is a lossy codec. It is able to get extreme savings in size, because it optimizes out parts of the sound that most humans don't hear. However, that same bandwidth that humans don't hear is exactly the bandwidth that the fax application uses to transmit valuable portions of the image. Therefore, the GSM codec is never going to be appropriate for sending faxes. Besides, if you need low bitrates for your IP connection, you're likely to experience delays in the fax negotiation -- which will probably result in a failed fax attempt. If you want to be able to send faxes in this way, then negotiate the fax at one end, and email the resulting TIFF to the other end. Added to FAQ and http://www.voip-info.org/tiki-index.php?page=Asterisk+fax Thank you both! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceMail Password problems
I'm having problems retrieving messages. 1 I dial the ext to run VoiceMailMain 2 VoiceMailMain asks for my password 3 I enter password (via BT-100 phone) on the keypad followed by #. 4. When examining the asterisk console I see some of the digits I entered for my passwd are repeated. For example if I enter password 4321#. The console shows Incorrect Password '4433211' for user '2000' (context =any) Any ideas why the digits enterd for the password are being repeated?? rgds burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail notification problem
When one exension has a voicemail, stutter dialtone is turned-on for all extensions. My zapata.conf entries: [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] My voicemail.conf entries: [other] 61 = 1234,...,[EMAIL PROTECTED] 62 = 1234,...,[EMAIL PROTECTED] etc. Are my entries correct, or is there a way to fix it so that just the extension with the voicemail gets stutter dialtone? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMail Password problems
Incorrect Password '4433211' for user '2000' (context =any) This is a FAQ: use dtmfmode=info in your sip.conf for your Grandstream Note: Don't forget to reload after modifying sip.conf. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ALSA use with Asterisk
Can anyone point me in the right direction? I'm trying to get asterisk to use my ALSA compatible sound card(configured correctly) and my Conexant modem to get on an outside line. Or can someone point me to where the documenation for teh ALSA conf is located? I felt competent wiht linux, but asterisk is prepping me for the funny farm. If I am missing soemthign, or you have anything tosay. Even if it's, Hey stupid you forgot to turn off noload=alsa and turn on noload=oss in teh module conf My aprreciation in advance; Haxot Dominant: Lonely Guy/Nanny/Archivist Recessive: Lurker/Therapist/Palooka What's YOUR genes? http://www.winternet.com/~mikelr/flame1.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unsubscribe
_ Add photos to your e-mail with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/photospgmarket=en-caRU=http%3a%2f%2fjoin.msn.com%2f%3fpage%3dmisc%2fspecialoffers%26pgmarket%3den-ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMail Password problems
Thanks a lot. It works fine now On Sat, 13 Dec 2003 21:27:21 +0100, Philipp von Klitzing wrote Incorrect Password '4433211' for user '2000' (context =any) This is a FAQ: use dtmfmode=info in your sip.conf for your Grandstream Note: Don't forget to reload after modifying sip.conf. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users rgds burak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] new CVS Checkout
On Sat, 2003-12-13 at 10:32, rnc Info Lists wrote: On Sat, 2003-12-13 at 16:41, Joe Dennick wrote: I just updated yesterday, but I did a complete rm -Rf for all of the following directories: /usr/src/zaptel /usr/src/zapata /usr/src/libpri /usr/src/asterisk Then I did a new cvs checkout for all four of those items before recompiling them (make clean; make install) in the same order. My Asterisk now states that its running Version CVS-12/12/03-09:47:51. I use make update. In the Makefile under update: is the instruction rm -f .version -- Dave Cotton [EMAIL PROTECTED] Thanks to everyone who replied to this question. I got rid of the hidden files and reloaded from CVS. It is ok now.Guess over the next days I'll go through the archives and learn a bit more about using CVS to maintain updates. While reading up on CVS is a good thing and I do not wish to dissuade you, I doubt you will find anything in there about the .version file. The .version file is just something created by the make file so you can track your version somewhat usefully. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMail Password problems
I don't get why people always say dtmfmode=info mine works fine with rfc2833. bkw On Sat, 13 Dec 2003, Philipp von Klitzing wrote: Incorrect Password '4433211' for user '2000' (context =any) This is a FAQ: use dtmfmode=info in your sip.conf for your Grandstream Note: Don't forget to reload after modifying sip.conf. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extension response
When asterisks gets and incoming call the server is setup to dial a zap phone and then plays a message telling the caller what extension to press to leave a message. During that message I can not press the voicemail extension or any other extension that is included. Thank in advance for any help, Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Call not transferred - plz help
I have a problem with IAX call transfer. The call goes successful but consumes lot of BW in the middle tier. The actual connection is like this (NAT) DIAX(IAX2) - *1 -- *2 *1 *2 were public IP with asterisk. It consumes around 120kbps in total to forward a single GSM call. I have the following configuration in my iax.conf [general] ... disallow=all allow=gsm [provider] type=peer username=userid secret=password host=myprovider.com i can successfully make a call to my provider through the following settings in extensions.conf exten = _X.,1,Dial(IAX2/[EMAIL PROTECTED]/extension|tr) but the call is still natively bridged. Can you help me how can i avoid native bridging. Accepting AUTHENTICATED call from 81.86.88.233, requested format = 2, actual format = 2 -- Executing Goto([EMAIL PROTECTED]/1, 13731|s|1) in new stack -- Goto (13731,s,1) -- Executing Wait([EMAIL PROTECTED]/1, 2) in new stack -- Executing Answer([EMAIL PROTECTED]/1, ) in new stack -- Executing DigitTimeout([EMAIL PROTECTED]/1, 3) in new stack -- Set Digit Timeout to 3 -- Executing ResponseTimeout([EMAIL PROTECTED]/1, 10) in new stack -- Set Response Timeout to 10 == CDR updated on [EMAIL PROTECTED]/1 -- Executing Dial([EMAIL PROTECTED]/1, IAX2/[EMAIL PROTECTED]/845745|40|tr) in new stack -- Called [EMAIL PROTECTED]/845745 -- Call accepted by 69.28.208.84 (format GSM) -- Format for call is GSM -- IAX2[provider]/4 is ringing -- IAX2[provider]/4 stopped sounds -- IAX2[provider]/4 answered [EMAIL PROTECTED]/1 -- Attempting native bridge of [EMAIL PROTECTED]/1 and IAX2[provider]/4 -- Channel 'IAX2[provider]/4' unable to transfer -- Hungup 'IAX2[provider]/4' Can anyone please help me where could be the problem and how to avoid native bridging. Thanks for your help. Kannaiyan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMail Password problems
Hi! I don't get why people always say dtmfmode=info mine works fine with rfc2833. bkw Dunno. I tried rfc2833 first, and had exactly the same problem as described below with voicemail (but only there). Info then worked just fine (as obviously also confirmed by this user here). Is there any other setup/setting that has influence on DTMF detection? Like NAT (yes for me) or anything else? However, more likely it's simply a GS firmware thing (4.17 on mine) - or production (hardware) issue with GS. Incorrect Password '4433211' for user '2000' (context =any) This is a FAQ: use dtmfmode=info in your sip.conf for your Grandstream Note: Don't forget to reload after modifying sip.conf. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extension response
On Sat, 2003-12-13 at 22:18, Matthew Pallotta wrote: When asterisks gets and incoming call the server is setup to dial a zap phone and then plays a message telling the caller what extension to press to leave a message. During that message I can not press the voicemail extension or any other extension that is included. Did you answer() the line? Are you using Playback() instead of Background()? Playback() does not accept DTMF. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
