[Asterisk-Users] Dropped calls if transferred across servers into MeetMe with mobile source
I have an application where calls come into an *box from a DID provider, and may be transferred to a meetme conference on another *box (the call is released by the first *box after transfer). These are ulaw IAX channel calls, and if the source is from a Verizon or Nextel mobile phone to the DID (other carriers not tested), the call drops about 2-3 minutes after it joined the meetme conference. POTS originated calls do fine - they do not drop. I've reproduced this consistently, and across two different DID termination providers and several different mobile phones. I'm seeing the behavior on 1.0.7 and 1.0.9. Calls don't fully drop. Meetme shows a reduction in the participant count, and the conference exit tone plays, but the mobile phone thinks it is still connected... AND other call participants can still hear the 'dropped' person, but that person can't hear anything. Also, if I change the first *box iax.conf to notransfer=yes, all calls are reliable (but, of course, I'm tying up resources...not a good long term solution). Console output is as follows for problem calls: Jul 5 15:18:44 WARNING[9256]: chan_iax2.c:1477 attempt_transmit: Max retries exceeded to host xx.xx.xx.xx on IAX2/yyy@ xx.xx.xx.xx:4569/3 (type = 2, subclass = 4, ts=65540, seqno=1) Jul 5 15:18:44 WARNING[9256]: app_meetme.c:962 conf_run: Unable to write frame to channel: No child processes == Spawn extension (toll, 1001074, 5) exited non-zero on 'IAX2/ yyy@ xx.xx.xx.xx:4569/3' -- Hungup 'IAX2/yyy@ xx.xx.xx.xx:4569/3' I've experimented with jitterbuffer on and off, different qos settings (including high reliability), and different meetme options. I haven't been able to impact this behavior. There is an agi that executes when the call arrives at the meetme *box (before meetme is joined). It just hits a db, sets some variable values, and exits cleanly - and again, it's not until 2-3 minutes later that I see the problem, and I don't have any problem with POTs sourced calls. The big variable seems to be whether the call originated from a cell phone or not, and that it was transferred to a second server. This is really strange, and I've even pulled in someone else that does Asterisk work just to do a sanity check and make sure I wasn't missing something obvious... no such luck. Any thoughts or insights? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Teliax Passing Audio?
Is anyone having issues with audio being passed inbound via Teliax? Trying to isolate an issue here. Thx, -Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with iax2 and 2 peers behind nat
That is not going to work. Asterisk shouldn't be behind a NAT to get registration of boxes behind NAT. I've done it, and it works. It is not a great situation though because of the provisioning problem. Specifically, an IAX device behind NAT has no way of getting its provisioning out of the blue from outside the LAN. If you provision from inside, no problem. Several clients should be able to register as log as they are using different login account info and of course different ports. The second unit will need to use a port other than 4569 and it should do so automatically. Even my $70 phones do this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk perl radiusclient
hello austin how to install perl module i m following http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth and i did sudo perl -MCPAN -e shell; install Config::IniFiles install Crypt::CBC install Crypt::DES install Authen::Radius any other help full link i m new to perl JD Austin wrote: It's complaining that you don't have the perl module installed or it is not in your path. Kamran Ahmad wrote: hello how to solve these errors /var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10 use Asterisk::AGI; vi /etc/asterisk/extensions.conf exten = _X.,1,agi,agi-rad-auth.pl|Routing=SIPAuthorizeBy=SIP vi /etc/asterisk/modules.conf load = res_agi.so ---errors *CLI Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.5/i386-linux-thread-multi /usr/lib/perl5/5.8.5 /usr/lib/perl5/site_perl/5.8.5/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.4/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.3/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.2/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.1/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.5 /usr/lib/perl5/site_perl/5.8.4 /usr/lib/perl5/site_perl/5.8.3 /usr/lib/perl5/site_perl/5.8.2 /usr/lib/perl5/site_perl/5.8.1 /usr/lib/perl5/site_perl/5.8.0 /usr/lib/perl5/site_perl /usr/lib/perl5/vendor_perl/5.8.5/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.4/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.3/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.2/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.1/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.5 /usr/lib/perl5/vendor_perl/5.8.4 /usr/lib/perl5/vendor_perl/5.8.3 /usr/lib/perl5/vendor_perl/5.8.2 /usr/lib/perl5/vendor_perl/5.8.1 /usr/lib/perl5/vendor_perl/5.8.0 /usr/lib/perl5/vendor_perl .) at /var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10. BEGIN failed--compilation aborted at /var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10. Jul 6 19:38:54 WARNING[30695]: app_dial.c:516 dial_exec: Dial argument takes format (technology1/number1technology2/number2...|optional timeout) Sell on Yahoo! Auctions no fees. Bid on great items. http://auctions.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Not MRTG, what about ARGUS?
I use Nagios to monitor lines. I use the check_asterisk script that you'll find floating around the place. I connect via the mgmt interface. Added to nagios is nagiosgraph. This keeps historical RRD graphs of my line usage. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Carlos Alperin wrote: Ok, I got it. None use MRTG to track status history on Asterisk. Someone uses ARGUS? Any other tool? Someone track their lines? HEL Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sccp new realease
Remco Barende ha scritto: Does this version of chan_sccp replace the version at sourceforge or is this Yet Another Fork(tm) :) It's a fork. Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Not MRTG, what about ARGUS?
On Wed, Jul 06, 2005 at 11:32:11PM -0400, Carlos Alperin wrote: That sound like the Spanish TV Show. Is a similar of MRTG? If that is the case, the problem is the SNMP module for Asterisk. Why use snmp? you don't weant to minitor asterisk's snmp. You want to monitor Asterisk. Either poll asterisk's voip ports (sip/iax/whatever) to ping it, or use the manager interface if allowed. The manager interface gives you a nice TCP port. MRTG uses SNMP for routers. Alternatively, it can use any script that returns a simple 4-lines output. See the the mrtg-contrib directory for details. 'asterisk -rx' can be used to generate output (after filtering out verbose/debug). What numbers exactly do you want to get? It was made for use with UCD-SNMP, not for NET-SNMP. And my platforms are all RH-9. That is why I never was able to made it work. ucd-snmp was renamed to net-snmp on version 5, I believe. Then, tired to look for a miracle I decide to try ARGUS, which suppose to bring the module for Asterisk by default. And guess what? Argus support list told me today: Oh, we never was able to make it work with Asterisk If works, I can install BIG FATHER if I need too. Big Brother is a non-free program, and as a result is pain to install. Still, you have to admire their choice of port numbers. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quadBRI form junghanns.net
Got a few and 8line one running good,got some compatibility problems with some mother boards once but that was it On Wed, 2005-07-06 at 16:08 -0300, Bartosz Jozwiak wrote: Hello, Is anybody there using quadBRI form Junghanns.net with Asterisk ? I would like to order that card but first would like to hear some opinions. Thank you in advance Bartosz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quadBRI form junghanns.net
I had quite a lot of experience with it ... it works fine, the only problem I got was that I couldn't transmit fax (data) calls through it reliably ... although this was some time ago, so it is possible that the kernel modules for them improved lately. Ivan Hello, Is anybody there using quadBRI form Junghanns.net with Asterisk ? I would like to order that card but first would like to hear some opinions. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: zaptel missing /dev/zap after FC3 update
In article [EMAIL PROTECTED], Howard Ratzlaff [EMAIL PROTECTED] wrote: I have a P4 HT system running * on a 2.6.9-1.667smp kernel (Fedora Core 3). Everythng was testing out and the configuration was working. After running YUM update, kernel 2.6.11-1.35_FC3smp was installed. Now Zaptel cannot find /dev/zap. Waiting for zap to come online...Error: missing /dev/zap! I have already recompiled zaptel, libpri, and asterisk after changing the /usr/src/linux-2.6 symbolic link (linux-2.6 - /lib/modules/2.6.11-1.35_FC3smp/build/). There is only a TDM22b installed I reverted to the older kernel, recompiled and have the same issue. Any thoughts? I have found that sometime the devices take a bit longer to become available and the init script gives up too soon. In /etc/rc.d/init.d/zaptel, try changing TMOUT=10 to TMOUT=20 Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Change Authorization to Proxy-Authorization
How can I change: Authorization: Digest username=70501956, realm=taraba.net, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=42ccd58240bd61c429ab1d2479d00209867a16a0,response=02fe9acd0bcb5f1866854b85439aebeb, opaque= to be: Proxy-Authorization: Digest username=2201,realm=taraba.net,nonce=45e12a1c,response=c862acc59c 3914311b52e1bad7f8f4a5,uri=sip:[EMAIL PROTECTED] Does anybody know? :-( I think I need a sip-proxy setting but I cannot find anything written on how to do this. Jason Frisch ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] DECT VoIP Gateway
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of IM.Nobody Sent: Wednesday, 6 July 2005 11:51 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] DECT VoIP Gateway Hi all, Just want to share with all of you a new hot DECT VoIP gateway available from www.broad-tel.com/index_en.php. The DECT VoIP gateway is capable of handling both SIP and the H.323 calls. Up to 4 registrations to the SIP proxy or H.323 Gatekeeper. To bring the users most flexibility, the add-on RJ-11 interface for PSTN connection, users not only can make the daily PSTN communication, but also enjoy the convenience brought by VoIP communications. With built-in DECT GAP Compatible base, up to 5 DECT handsets can be registered on the gateway. Cheers, IM Is it just me that sees the post above as spam? If we (tinw) even consider buying stuph from spammers, then we are encouraging them in their sociopathic behavior, and as a consequence they will do more spamming. What is the consensus here? I would have found it interesing *if* I could find any information on the website about the product? Am I blind or where is it? /Ola ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any SIP hardphone recommendations?
or ci$co 7940 - features same as 7960, but only with two lines, instead of six, but significantly cheaper than 7960... PJ Glenn Powers wrote: Cisco 7960's work well and are highly recommended by many people, including myself. They have the qualities you list. cheers, glenn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Senao WiFi SIP Phone SI-680H
Hi, Have anyone succesfully configured wifi roaming using Senao Wifi phone model SI-680H? If yes, please let me know your phone's firmware version and your configuration. Thank you. -eddie- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with Cisco 7905G corrupted image!!
If you are end user, there will be problem with direct communication with ci$co, because ci$co standard way to solve problem is via ci$co partner/reseller that sell the phone to you :-( PJ Andres Maduro wrote: Hi, I recently purchased from a friend 2 Cisco 7905G for testing them with Asterisk. I was able to upgrade one of them with the SIP image, the other hang up during the upgrade process and now it won't boot again. When powered up, the red and green lights keep on and the screen is blank. Does any one know a procedure to fix this ? I do not have a contract with Cisco, I have even call a friend in Cisco which said that I must purchase a contract to be able to open a TAC ticket. Any help is greatly appreciated. Regards. AM. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quadBRI form junghanns.net
Ivan Meic (Vox Mundi) wrote: I had quite a lot of experience with it ... it works fine, the only problem I got was that I couldn't transmit fax (data) calls through it reliably ... although this was some time ago, so it is possible that the kernel modules for them improved lately. I can confirm the 'sending' is a bit problematic indeed. (on SuperMicro mainboard, no other issues) Last time we tested was, eh, a few days ago :-) Receiving works perfect. Cheers, Kristof. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Dialplan configuration with Realtime
snacktime wrote: On 7/6/05, Gundemarie Scholz [EMAIL PROTECTED] wrote: Following the instructions on voip-ip.org I have implemented Realtime with MySQL for my Asterisk server. The individual extension configuration is managed in a table called extensions. Still I have to keep some data in the extensions.conf, namely the switch and the include statements. Is there a way to minimize that or completely get rid of them? No, but you can put extensions.conf into mysql via realtime static I have done so, and for testing I have added the switch and include statements from extensions.conf, after that I have removed the extensions.conf file completely. Here is an excerpt of the ast_config table, shortened to the relevant bits. ++-+-+--+--+-+ | id | cat | var | category | var_name | value | ++-+-+--+--+-+ | 6 | 0 | 0 | general | static | yes | | 7 | 0 | 1 | general | writeprotect | no | | 38 | 1 | 0 | default | switch | Realtime/[EMAIL PROTECTED] | | 39 | 2 | 0 | from-sip | include | default | | 40 | 3 | 0 | local| include | default | ++-+-+--+--+-+ During startup Asterisk then unfortunately produces the following error message: WARNING: pbx.c:3650 ast_merge_contexts_and_delete: Requested contexts didn't get merged. while using realtime extensions at the same time. I am doing that already, and it works well as long as I use switch and include statements in an extensions.conf file. If your goal is to keep everything in the database that will work. The question then remains where to put the include and the switch statements, as they don't seem to fit into the extensions table either. Any suggestions are highly appreciated! mysql show columns from extensions; +--+--+--+-+-+---+ | Field| Type | Null | Key | Default | Extra | +--+--+--+-+-+---+ | context | varchar(20) | | PRI | default | | | exten| varchar(64) | | PRI | | | | priority | int(2) | | PRI | 1 | | | app | varchar(20) | | | | | | appdata | varchar(255) | YES | | NULL| | | descr| text | YES | | NULL| | | flags| int(1) | | | 0 | | +--+--+--+-+-+---+ Regards, Gunde ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and wireless on site personal paging system
hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. the customer is currently using a paging system (small receivers which display a callback number and a base station (transmitter) with several antennas at the site) the problem is, that the currently operative base station uses 4 ISDN BRI interfaces. But the protocol is old germany 1TR6 (and not EuroISDN). - is there anybody with experience on these pager devices? do they have a common standard? - does anybody know of a pager base station with an SIP interface? - does anybody know of a pager base station with an EuroISDN interface? what's your general advice on those paging systems? regards frank sautter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] experience with analog channel banks in E1 land
hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. it will be a slow migration, the asterisk server will be inserted between the telco E1 and the hicom. new phones will be sip ones. the customer has several fax machines and analog phones (some of them have to be explosion-proof). around 50 analog ports in total are needed. as we are in E1 land (germany) we have 64kpbs per channel. most (affordable) channels banks are T1 (56kpbs per channel i assume). the questions are: - could the T1 channelbanks be connected to a TE405P with two channels in E1 mode (telco and hicom pbx) and two channels to the channel banks (i think yes, but just to be shure)? - will the faxmachines work (56kpbs-64kbps)? is asterisk translating this (btw. how do faxes work from europe to north america - the telcos have the same problem)? - which signalling protocoll will be used on the T1 side? is asterisk translating this correctly? - btw. where is the different bitrate coming from? is it 7bit T1 and 8bit E1 or 7kHz and 8kHz sample rate? regards frank sautter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [Asterisk-Users] DECT VoIP Gateway
http://www.broad-tel.com/products/wireless.php On 7/7/05, Ola Lidholm [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of IM.Nobody Sent: Wednesday, 6 July 2005 11:51 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] DECT VoIP Gateway Hi all, Just want to share with all of you a new hot DECT VoIP gateway available from www.broad-tel.com/index_en.php. The DECT VoIP gateway is capable of handling both SIP and the H.323 calls. Up to 4 registrations to the SIP proxy or H.323 Gatekeeper. To bring the users most flexibility, the add-on RJ-11 interface for PSTN connection, users not only can make the daily PSTN communication, but also enjoy the convenience brought by VoIP communications. With built-in DECT GAP Compatible base, up to 5 DECT handsets can be registered on the gateway. Cheers, IM Is it just me that sees the post above as spam? If we (tinw) even consider buying stuph from spammers, then we are encouraging them in their sociopathic behavior, and as a consequence they will do more spamming. What is the consensus here? I would have found it interesing *if* I could find any information on the website about the product? Am I blind or where is it? /Ola ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quadBRI form junghanns.net
howdy, the problems with data and fax calls were mainly caused by asterisk, e.g. echo cancelation always on, failed native bridging, gains, Since bristuff 0.2.0-RC8e those issues have been solved. We have quite a few customers running loads of ISDN data calls between their locations without any special asterisk options. best regards Klaus -- Klaus-Peter Junghanns Am Donnerstag, den 07.07.2005, 09:13 +0200 schrieb Ivan Meic (Vox Mundi): I had quite a lot of experience with it ... it works fine, the only problem I got was that I couldn't transmit fax (data) calls through it reliably ... although this was some time ago, so it is possible that the kernel modules for them improved lately. Ivan Hello, Is anybody there using quadBRI form Junghanns.net with Asterisk ? I would like to order that card but first would like to hear some opinions. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_conference and AGI
At 15:21 06/07/2005 +0200, Tobias Wolf wrote:
Hi,
i was successful in compiling app_conference and setting up an conference
was quite easy. :-)
Does anyone knows if it is possible to have an IVR accessable from inside
the conference. So, if i dialed into an conference i want to be able to
press '*' and then the actual discussion is muted for me and i and menu is
read to me. Something like the ${MEETME_AGI_BACKGROUND} in MeetMe.
Thx in advance :)
Tobias Wolf
Looking at apps/app_meetme.c, I saw that there is a POUNDEXIT option
that when set will kick a user when she hits #, you use 'p' as an option
when invoking Meetme in the dial plan.
There is another option, STARMENU, that enables an admin menu when user
hits * ('s' option)
I guess that you could either change your mind and use # or patch
app_meetme to accept both # and * (when STARMENU is not enable) or
patch app_meetme to inverse the roles of # and *. Ideally you want both
DTMFs to be configurable instead of hard coded, but that's another story.
Once you get what you want there, i.e. the ability to leave the conference,
you will handle the IVR in the dialplan I suppose. When done, you get back
to the conference room in meetme (assuming you tracked it).
But... what is wrong with MEETME_AGI_BACKGROUND ? (besides the fact
that unfortunately it does not work for SIP channels due to the mixing
not being done in the zaptel driver but app_meetme itself, sort of, AFAIK).
Hope this helps,
Yours,
JeanHuguesRobert
-
Web: http://hdl.handle.net/1030.37/1.1
Phone: +33 (0) 4 92 27 74 17
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Re: [Asterisk-Users] OT: Congrats, Europe!
Vahan Yerkanian wrote: http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136 http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/ So we're are waiting the free g729 codec for Europe now ... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] experience with analog channel banks in E1 land
Hi! - could the T1 channelbanks be connected to a TE405P with two channels in E1 mode (telco and hicom pbx) and two channels to the channel banks (i think yes, but just to be shure)? Yes - no problem. - will the faxmachines work (56kpbs-64kbps)? is asterisk translating this (btw. how do faxes work from europe to north america - the telcos have the same problem)? Usually the fax is analog 14,4k connection. The channelbank does the digitizing to ulaw and the other end coverts it back to analogue. Afaik no matter if its european or american ISDN. No problems experienced by now. - which signalling protocoll will be used on the T1 side? is asterisk translating this correctly? That depends on your channel bank. Rhino channel banks for example are specially designed for asterisk. But most T1 channel banks should do. - btw. where is the different bitrate coming from? is it 7bit T1 and 8bit E1 or 7kHz and 8kHz sample rate? Don't kow that ;-) Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quadBRI form junghanns.net
Klaus, Can the data transmission work reliably now between an incoming PRI line (Digium TE405P) and outgoing BRI line (QuadBRI) ? Ivan the problems with data and fax calls were mainly caused by asterisk, e.g. echo cancelation always on, failed native bridging, gains, Since bristuff 0.2.0-RC8e those issues have been solved. We have quite a few customers running loads of ISDN data calls between their locations without any special asterisk options. best regards Klaus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quadBRI form junghanns.net
Ivan, as long as you use BRIstuff it will work fine with any zaptel hardware, even with Digium or Sangoma. best regards Klaus -- Klaus-Peter Junghanns Am Donnerstag, den 07.07.2005, 12:25 +0200 schrieb Ivan Meic (Vox Mundi): Klaus, Can the data transmission work reliably now between an incoming PRI line (Digium TE405P) and outgoing BRI line (QuadBRI) ? Ivan the problems with data and fax calls were mainly caused by asterisk, e.g. echo cancelation always on, failed native bridging, gains, Since bristuff 0.2.0-RC8e those issues have been solved. We have quite a few customers running loads of ISDN data calls between their locations without any special asterisk options. best regards Klaus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Congrats, Europe!
On Thu, 7 Jul 2005, Anton Tinchev wrote: Vahan Yerkanian wrote: http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136 http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/ So we're are waiting the free g729 codec for Europe now ... No need for celebration... http://www.cnn.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and wireless on site personal paging system
On Thu, 2005-07-07 at 11:21 +0200, Frank Sautter wrote: [snip] the problem is, that the currently operative base station uses 4 ISDN BRI interfaces. But the protocol is old germany 1TR6 (and not EuroISDN). Did you try contacting the vendor of the base stations to see if they have a EuroISDN firmware update? My Eicon Diva Server BRI card supports the 1TR6 protocol. The firmware can be found here: ftp://ftp.isdn4linux.org/pub/isdn4linux/utils/eicon/firmware/ Perhaps AVM supports 1TR6 too. - does anybody know of a pager base station with an SIP interface? - does anybody know of a pager base station with an EuroISDN interface? Nope. Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] disconnect with various codecs
Hello list, I'm pretty new to Asterisk, but so far I managed to setup the server, added a couple of mISDN channels (one TE, one NT), connected to a VOIP provider and called out to the World :) Now I started to play with codecs because I wanted to try the sound quality of each of them, and something weird happens. My setup is to check one codec at a time, so I disallow all, then allow one. In iax.conf: [general] disallow=all allow=ilbc [channeldefs...] Now when I try to call, the VOIP provider sets the connections through (i.e. my phone is ringing), but once I lift the receiver, the VOIP provider hangs up the connection: mISDN--- Asterisk -- VOIP provider -- POTS ringing Once I lift the POTS handset, the connection between VOIP provider and POTS closes (so I only hear a busy tone). What could be happening here? Best regards, Valentijn -- http://www.openoffice.nl/ Open Office - Linux Office Solutions Valentijn Sessink [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Teliax Passing Audio?
Is anyone having issues with audio being passed inbound via Teliax? Trying to isolate an issue here. Nope, works fine here with cvs-head from about a week ago. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calls with oh323 with no sound
Hi, I've oh323 chan installed and working to make calls from SIP to H323 devices. The problem is can no hear sound with the H323 device. I think this is some related with codecs o nat, because the H323 have one public IP from a different subnet from the asterisk box. If I use netmeeting in gateway mode, the call can be completed and I can talk with a SIP device, but in gateway mode I can not call netmeeting from SIP device. This is the oh323.conf : ; Configuration file of OpenH323 channel driver ; ;- ; General configuration options ; (ports, jitter, GK, ...) ;- [general] ; ; Address to bind to for incoming connections. ; Default is ALL. ; listenAddress=0.0.0.0 ; ; Port to listen to. ; Default value is 1720. ; listenPort=1720 ; ; Port to connect to. ; (Used only when we don't have a gatekeeper) ; Default value is 1720. ; connectPort=1720 ; ; Configure TCP port range to be used by H.323 ; tcpStart=1 tcpEnd=2 ; ; Configure UDP port range to be used by H.323 ; Note: The port range used by RTP are configured from ; rtp.conf ; udpStart=1 udpEnd=2 ; ; Enable fast start (yes,no). ; fastStart=yes ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=no ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=yes ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; inBandDTMF=no ; ; Enable silence suppression. ; silenceSuppression=no ; ; Set jitter buffer (in milliseconds, 20...1). ; jitterMin=20 jitterMax=100 ; ; Set IP Type-of-Service byte for RTP channels. ; Valid values for this option are: ; lowdelay, throughput, reliability, mincost, none ; ipTos=lowdelay ; ; Set the maximum number of inbound/outbound/simultaneous ; H.323 connections. ; outboundMax=10 inboundMax=10 simultaneousMax=10 ; ; Set the bandwidth limit for H.323 connections. ; The value is in Kbps. ; ;bandwidthLimit=1024 ; ; Set tracing options for the wrapper library and for the ; OpenH323 library. ; libTraceFile can be 'stdout' or a full path name to the tracefile. ; Only trace info for OpenH323 is logged in libTraceFile. ; wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout ; ; Disable gatekeeper or specify a gatekeeper. ; Valid values for this option are: ; DISABLE, ; DISCOVER, ; gatekeeper's DNS name, ; gatekeeper's ip, ; GKID:gatekeeper's id ; ;gatekeeper=192.168.1.2 gatekeeper=DISCOVER ; ; Set the gatekeeper password ; ;gatekeeperPassword=secret ; ; Set the gatekeeper registration timeout ; gatekeeperTTL=600 ; ; Set the mode for sending user-input ; Valid values for this option are: ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 ; userInputMode=RFC2833 ; ; AMA flags (default, omit, billing, documentation) ; amaFlags=default ; ; Account code ; accountCode=H323 ; ; Set the default context of H.323 calls. ; ;context=voip-h323 ;context=from-pstn context=from-internal ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; Colocar las extensiones SIP en esta seccion alias=asterisk ; Para el Voice Mail alias=*98 ; Los teléfonos alias=100 alias=101 alias=102 alias=103 alias=104 alias=105 alias=106 alias=107 alias=108 alias=109 alias=110 alias=200 alias=201 alias=202 alias=203 alias=204 alias=205 alias=206 alias=207 alias=208 alias=209 alias=210 alias=500 alias=501 alias=502 ; ; Aliases/prefixes routed in all-aliases context. ; context=all-aliases alias=ASTERISK alias=666 ; ; Aliases/prefixes routed in more-aliases context. ; context=more-aliases alias=665 ; ; Aliases/prefixes routed in all-prefixes context. ; context=all-prefixes gwprefix=00 gwprefix=01 ; ; Aliases/prefixes routed in more-stuff context. ; context=more-stuff alias=664 gwprefix=02 ;- ; Specify and configure CODEC related ; options ;- [codecs] ; ; Define the codec list of the channel driver. ; Every codec option may have a frames option ; associated with it. ; Valid values for the codec option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3- G.723.1(6.3k) ; G72315K3- G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726- G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728- G.728 ; G729- G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP
[Asterisk-Users] Can I connect to an existing network?
I am new to asterisk and have a few architecture questions. I currently have a 3MB bonded T1 running our network and was wondering if Asterisk be connected to the existing network and bonded T1 (which also includes normal day to day network traffic), or do I have to dedicate a new T1 to asterisk? Also, do I need to use PRI lines, or can all voice calls be sent over the T1 line? Are there any issues I should be aware of when implementing an Asterisk server and also what hardware would you recommend for my situation? Thank you for your help, Jyran Glucky Advisory Programmer BlueWare, Inc. Strategic HealthWare Solutions 3060 W. 13th Street Cadillac, MI 49601 Phone: (231) 779-0224 ext. 111 Fax: 231-779-1002 mailto:[EMAIL PROTECTED] http://www.blueware.net DID YOU KNOW? BlueWare is the Grand Prize Winner of the 2005 IBM Beacon Award BEST DB2 (Document Management) Application Worldwide. BlueWare Market Share for Hospital Document Management Systems is in 25 states in the US. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues and busy agents problem
Hi
I have a problem with the queues on Asterisk. The setup is [EMAIL PROTECTED]
v1.0 with Asterisk 1.0.7.
I have 1 queue (4500) set up, with leastrecent strategy. There are no
agents configured in this queue.
Agents log in by dialing 4500* on their phones. All incoming calls are sent
to the queue. Calls wait 120 seconds in the queue, and are then sent to
voicemail extension 310.
My problem is that while an agent is busy on a call, Asterisk is still
sending calls to that agent.
The queue configuration from extensions.conf is:
[ext-queues]
exten = 4500,1,Answer();
exten = 4500,2,SetCIDName(${CALLERIDNAME});
exten = 4500,3,Queue(4500|t|||120); Support
exten = 4500,4,Macro(vm,310);
exten = 4500*,1,Macro(agent-add,4500,);
exten = 4500**,1,Macro(agent-del,4500);
[macro-agent-add]
exten = s,1,Wait(1)
exten = s,2,GotoIf($[foo${ARG2} = foo]?4:3))
exten = s,3,Authenticate(${ARG2})
exten = s,4,AddQueueMember(${ARG1})
exten = s,5,Wait(1)
exten = s,6,Playback(agent-loginok)
exten = s,7,Hangup()
[macro-agent-del]
exten = s,1,Wait(1)
exten = s,2,RemoveQueueMember(${ARG1})
exten = s,3,Wait(1)
exten = s,4,Playback(agent-loggedoff)
exten = s,5,Hangup()
The queue configuration from queues.conf is this:
[4500]
wrapuptime=0
timeout=20
strategy=leastrecent
retry=5
music=default
maxlen=0
leavewhenempty=yes
joinempty=yes
announce-holdtime=no
announce-frequency=0
agentannounce=None
Has anyone had a similar problem on Asterisk? I can't figure out why
app_queue doesn't know that the agent is busy on a call.
Regards
Hilton
Datatex Dynamics CC
Web site http://www.datatex.co.za/
Email to [EMAIL PROTECTED]
Tel +27215924033
Fax +27215924077
The use of the Datatex e-mail facility is not permitted
for the distribution of chain letters or offensive email
of any nature whatsoever. Datatex hereby distances itself
from and accepts no liability in respect of the
unauthorised use of its e-mail facility or the sending of
e-mail communications for other than strictly business
purposes. Datatex furthermore disclaims liability for any
unauthorised instruction for which permission was not
granted. Any recipient of an unacceptable communication,
a chain letter or offensive material of any nature is
requested to report it to [EMAIL PROTECTED]
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Re: [Asterisk-Users] Re: Remote SIP Connections
You can try to open up port for SIP 5060udp and RTP 10-2udp... (default setting) to your asterisk box. You will also have to specify that your extensions are nat=yes your externip=xxx.xxx.xxx.xxx (in SIP.conf) so that the SDP protocol will write the public IP and port translations for RTP (voice data). If this doesn't work, switch to IAX2 protocol- there are many hard-phones out there that support IAX2 protocol- You will only have to open up 4569udp on your firewall to your asterisk box and thats it. I have given my relatives an IAX2 hardphone so that we can all communicate... everything works well... (plus- I didn't have to configure or troubleshoot their firewall...major time saver!!!). Before you buy a Hardphone- try using an IAX2 softphone and see how it does for you... you can download one here: http://www.laser.com/dante/diax/diax.html cheers, francis On 7/6/05, Blake Krone [EMAIL PROTECTED] wrote: forgot to include the list -- Forwarded message -- From: Blake Krone [EMAIL PROTECTED] Date: Jul 6, 2005 9:07 PM Subject: Re: [Asterisk-Users] Re: Remote SIP Connections To: dbruce [EMAIL PROTECTED] Just had my brother connect from his time warner cable in minnesota to my adelphia in colorado springs, both NAT'd and I have my DMZ on, still nothing :( Any other ideas??? I wanted to setup an asterisk server so I could have VoIP in the house but then send SIP phones to my parents in Minnesota to save on long distance costs and cell minute usage. Thanks! On 7/5/05, Blake Krone [EMAIL PROTECTED] wrote: Well I had it setup with DMZ and port forwarding, removed the port forwards and still no luck :( Might end up going back to @home seen as other things like music on hold won't work properly, maybe something is just messed up in my gentoo install of asterisk. -Blake On 7/5/05, dbruce [EMAIL PROTECTED] wrote: You have forgotten that the WRT54G is a NAT router. The phones that are trying to connect to your server are also very likely to be behind a NAT router. This make it almost impossible to tell what ports are actually going to be used for inbound or outbound traffic... many NAT routers do not attach any significance to SIP protocol messages. Add to that the fact that many IP phones do not use the same port range for RTP that asterisk uses by default, and you have a VERY difficult time determining which port ranges need to be forwarded. Your easiest solution is to remove the forwarding rules, give your asterisk server a static IP address on your local network, and configure that IP address as the DMZ. All unsolicited requests to the router are sent to the IP address configured as the DMZ. The DMZ settings are found under the Applications Gaming tab on the WRT54G. You could also play with port triggering settings, but that is also a very dificult process. Regards, Derek Bruce - Original Message - From: Blake Krone [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, July 05, 2005 7:10 PM Subject: [Asterisk-Users] Re: Remote SIP Connections I have gotten them to be able to connect but I am unable to hear the other person and they can't hear me either. What else am I missing? On 7/5/05, Blake Krone [EMAIL PROTECTED] wrote: Hello all, I have my * server setup behind a Linksys WRT54G on Adelphia cable. I have forwarded 5060,1-10020, and another port set can't remember off the top of my head but I can't seem to connect to the * server from any locations that are direct connects to the Internet. Am I missing a portset for forwarding? If I use the name service (voip.*.com) from my home connection on the same LAN as the * server it will connect fine. Any ideas? TIA! -blake ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Francis Ballares [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.1
Our BIGGEST problem is that every single one of the 80 phones are on a direct connect T1. All are on qualify=1 yet sometimes we get 'TOO LAGGED'. HTF can you get that kind of lag on a dedicated, direct conected T1? Sounds more like a lost packet rather than lag... Try a ping -c 1000 192.168.10.10 and see what the max ping time is, and what the lost packets value is. Also, if that is happening mostly at the same time of day, then it is possible that you are sitting in some queue rather than being lost (I've seen ping times exceeding 8 seconds on an DSL connection), if that is the case, then you need to look at QoS. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mISDN transferring a call
Hello, Is it possible for an mISDN channel to transfer a call to a new phone, instead of opening a new channel to connect it? I have a couple of isdn phones connected to Asterisk; transferring a call will open a *second* isdn channel instead of connecting the two ISDN phones directly. Best regards, Valentijn -- http://www.openoffice.nl/ Open Office - Linux Office Solutions Valentijn Sessink [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_rxfax and app_txfax - Asterisk CVS HEAD
Hi all and specially Steve... Im using CVS HEAD with spandsp FAX solution. Im getting this erros when starting Asterisk: Jul 7 10:29:45 VERBOSE[31091] logger.c: [app_txfax.so]Jul 7 10:29:45 WARNING[31091] loader.c: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: fax_set_header_info Jul 7 10:29:45 WARNING[31091] loader.c: Loading module app_txfax.so failed! Any clues? Thanks. Denis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using G729 in pass through mode
Is it possible to use G729 on asterisk without the license? It is to connect devices which use the codec to termination providers in a phone card application. Will decoding the DTMF tones from the caller require G729 processing? This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Xten eyeBeam Video Problems
Blake Krone wrote: Hello all, I HAD video working before I upgraded to 1.08 (latest stable with Gentoo) and now it won't work. I just see noise bars and not the video. I know the camera works as I can use it in other programs such as AIM Yahoo. Which codec are you using for video in the eyeBeam? We have video IVR, voicemail, billing for video calls etc working fine here with multiple hardware and also the eyeBeam. My recommendation would be to allow only one video codec at a time in eyeBeam's confs. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Not MRTG, what about ARGUS?
I would like to check the status of my PRI's (I believe that should include ZAP), IAX (between Asterisk boxes) and SIP channels. The reason for choose MRTG was because they track historic using RRDB in a very good way. The reason for snmp, was that there was a snmp module developed for the Zapata by Andrea Fino, but she keeps announcing that only works for UCD-SNMP, not for NET-SNMP. Then I tried with UCD-SNMP on RH-9 but it never worked. At this time, I don't care what to use if I can see my status, by the way the traffic. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Thursday, July 07, 2005 2:56 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Not MRTG, what about ARGUS? On Wed, Jul 06, 2005 at 11:32:11PM -0400, Carlos Alperin wrote: That sound like the Spanish TV Show. Is a similar of MRTG? If that is the case, the problem is the SNMP module for Asterisk. Why use snmp? you don't weant to minitor asterisk's snmp. You want to monitor Asterisk. Either poll asterisk's voip ports (sip/iax/whatever) to ping it, or use the manager interface if allowed. The manager interface gives you a nice TCP port. MRTG uses SNMP for routers. Alternatively, it can use any script that returns a simple 4-lines output. See the the mrtg-contrib directory for details. 'asterisk -rx' can be used to generate output (after filtering out verbose/debug). What numbers exactly do you want to get? It was made for use with UCD-SNMP, not for NET-SNMP. And my platforms are all RH-9. That is why I never was able to made it work. ucd-snmp was renamed to net-snmp on version 5, I believe. Then, tired to look for a miracle I decide to try ARGUS, which suppose to bring the module for Asterisk by default. And guess what? Argus support list told me today: Oh, we never was able to make it work with Asterisk If works, I can install BIG FATHER if I need too. Big Brother is a non-free program, and as a result is pain to install. Still, you have to admire their choice of port numbers. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] isdn30 / pri lines in the UK
anybody recommend a supplier in the UK for a pri/isdn30 line (other than BT) thanx very much __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Q]: Asterisk + gnugk + BRI ISDN as H.323/ISDN gateway?
Hi I am currently using gnugk (www.gnugk.org) as a H.323 gatekeeper/proxy, mostly for our video conferencing devices. What I would now like to do is add ISDN functionality to this and make our H.323 gatekeeper box also function as an ISDN --- H.323 gateway so that H.323 endpoints can call ISDN devices, and vice versa. I currently have 3 BRI ISDN lines, but in order to use them I have to connect them to a specific video conf unit which supports ISDN (not all of our VC units do) and I also need to repatch the lines to wherever the unit is neededI would like to eliminate this by plugging the BRI lines into the gateway and making ISDN connectivity available to all our VC units (both ISDN and non ISDN units.they all support H.323). I had thought of using asterisk to take incoming ISDN calls and relay them as H323 to our gatekeeper/proxy. I was just wondering if anyone could help me with the following: 1. Is this possible with Asterisk + gnugk? 2. If so, can anyone recommend a multiport (preferrably 4 ports) BRI ISDN card which will work with asterisk? 3. Can asterisk do channel bonding so that multiple BRI ISDN can be used together for a video conference? Would this channel bonding work in both directions (i.e. H.323 -- ISDN and ISDN -- H.323)? If anyone has done this and has a working config, I would be most grateful for a copy. Thanx muchly in advance for any help. Chris Bradshaw. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: zaptel missing /dev/zap after FC3 update
Tony Mountifield wrote: In article [EMAIL PROTECTED], Howard Ratzlaff [EMAIL PROTECTED] wrote: I have a P4 HT system running * on a 2.6.9-1.667smp kernel (Fedora Core 3). Everythng was testing out and the configuration was working. After running YUM update, kernel 2.6.11-1.35_FC3smp was installed. Now Zaptel cannot find /dev/zap. What I've seen in the past is that if you do an upgrade (via Yum etc) that your udev rules and permissions will be saved as backups and new ones will be created. You can either copy the old ones back over the top of the new ones or just add the lines again (you can find the info to add in /usr/src/zaptel/udev/zaptel.rules and zaptel.permissions. The files you need to change/check are: /etc/udev/rules.d/50-udev.rules and /etc/udev/permissions.d/50-udev.permissions (the old ones would have .rpmsave appended) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DECT VoIP Gateway
Richard Malcolm-Smith wrote: Is it just me that sees the post above as spam? If we (tinw) even consider buying stuph from spammers, then we are encouraging them in their sociopathic behavior, and as a consequence they will do more spamming. What is the consensus here? It is a product announcement for a new product that is of intrest to users of asterisk, therefore not untargeted spam (like viagra etc mails) - I'm fine with it. That's exactly what the -biz list is for. Product announcements that may be of interest to Asterisk users. This is not a market place but rather a forum for the technical discussion of Asterisk. I have to agree that this is spam and should not be followed up. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using G729 in pass through mode
Hi, If you are terminating the call from/to a T1/E1 card or modifying the call in anyway e.g. playing IVR prompts not just voice in - voice out, you will require the codec. Regards, Sahil Gupta VoiceValley On Thu, 7 Jul 2005, Obelix wrote: Is it possible to use G729 on asterisk without the license? It is to connect devices which use the codec to termination providers in a phone card application. Will decoding the DTMF tones from the caller require G729 processing? This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] User proxy in SIP host
I am trying to get * to use proxy-auth when dialing out, to mimc what x-lite does when force proxy is set to yes. Is there any options that can be set to do this? This particular sip provider does not support the username:[EMAIL PROTECTED]/number for Dialing, so I connect as a peer. But it seems that * used www-auth in this case. If I can tell * that the host is actually a proxy it should use proxy-auth instead. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk/Grandstream Budgetone disconnect issue
Title: Asterisk/Grandstream Budgetone disconnect issue I am setting up a small Asterisk system for use at home. I have one Budgetone 101, one Cisco 7960 and two Xten lite softphones. So far everything is working expect for an issue with the Budgetone. When a call is placed between the Budgetone and any other phone, the call is setup and sounds good. If I hang-up on the Budgetone, everything is ok. If I hang up on the other phone, the Budgetone give me a busy signal, and does not hang up. Nothing is showing up in the messages log. Calls between the other phones work ok. Any ideas? Thanks. - A.G. Edwards & Sons' outgoing and incoming e-mails are electronically archived and subject to review and/or disclosure to someone other than the recipient. - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to read dbm or voltage via ztmonitor ?
Yes, I've read that. Ztmonitor is simply a very _basic_ tool that provides you with a little bit of feedback to adjust the rxgain and txgain settings to something relatively close to what the human ear considers reasonable audio levels. The tool cannot detect or determine what settings are reasonable for echo. Therefore, the end result is you need to listen to a conversation Isn't getting your transmit and receive levels to 0dBm considered THE #1 STEP in combating echo? The echo cancellers all assume that the audio will be at a given level and thus it's crucial to make sure this adjustment is fine. Sort of. Essentially the problem goes something like this... Audio transmitted from the CO incurres an unknown loss due to the length of the copper pstn lines between the CO and *. The longer the copper pairs, the greater the loss. If you're 15k feet from the CO, the loss is roughly 7db (the exact loss is dependent upon the gauge of the copper wires). So to get close to what another sip phone would sound like, one would need to crank rxgain=7.0. If you want the remote pstn party to hear properly, then txgain=7.0 is required as well. Setting both to that level causes echo as the levels are way outside what the * canceller can handle. Using that same example, if one could follow transmission engineering practices in use since the dark ages, the correct way to handle the loss would be to adjust the transmit level at the CO to be roughly 7db higher then normal so as deliver an audio signal to * that is roughly 0db (after the cable loss). And, then adjust * to transmit audio 7db greater then normal towards to CO. (In actual transmission engineering practice, the gains would typically be specified about 1 or 2db less then the cable loss.) However, I don't know of any telco that wouild do that even if they could, and I don't know of any current CO switch that has that capability on an ordinary pstn line-interface basis. If one think's about setting rxgain and txgain to values like 7db, think also about the noise, hum, and other imperfections that will also be amplified by the same 7db. Amplifying that crap also has a serious negative impact on how well the existing * canceller functions. So, essentially one can truly say the further * is from the CO, the greater the audio problems will be. In the olden days of US analog pbx trunking, the above example would be handled by the telco by using 4-wire analog circuits, and typically those circuits were engineered with the capability to adjust transmit levels at both ends of the circuit. Not so today with ordinary pstn lines. In this 4-wire approach, the echo canceller has much less work to do as the reflected energy that it has to deal with is much much less regardless of where it originates from. It would be very consistent with historic transmission engineering practices to essentially say use a TDM card if * is located within 3db of the CO, or use a digital (eg, pri/bri) line if beyond 3db. But, digium does not suggest/specify parameters like that and as a result we get lots of * implementors trying to use the TDM card in situations that are almost impossible to address. (Plus, as we've all seen over the last couple of years, the majority of * implementors don't have a clue what their pstn line loss happens to be or even how to measure it. Even those that represent themselves as professional pbx vendors don't typically own a transmission test set or know how to use a CO milliwatt generator.) Perhaps I have just been lucky but after adjusting the levels such that they're at 0dBm my echo problems are largely gone. Don't know if that's luck or just implementing asterisk with pstn lines that don't have relatively high cable losses. The US telco's have deployed a ton of digital remote line units (or cabinets) in our neighborhoods that effectively moves the CO line interface closer to our businesses and homes (less pstn loss). In those cases, we might think we're some hugh distance from the CO, but the remote line cabinets place the interface much closer to us then one might think. The average telco employee that rolls around in a truck and visits customer locations doesn't have a clue why those cabinets are out there. If I crank up the gain to compensate for a 7dB loss would the far end not get audio at correct levels? The 15kft isn't a moving target so that loss should be constant. Now if the hybrid on the TDM card is causing such jacked-up audio to be crossed over to the rx side you would *still* only see it as a sidetone since it's occurring right on the card and your only delays would be the digitization and PCI transfer delays. Your analogy with sidetone is correct, except with the TDM card there is an internal delay associated with transmitting data from asterisk (as an application) across the pci bus, queued on the TDM card, hybrid, receiving queue, pci bus, back into asterisk, and into * echo canceller. That
Re: [Asterisk-Users] Teliax Passing Audio?
I do have a problem see my post few yesterday with subject: Incoming 800-number over IAX -- #Joseph On Wed, 2005-07-06 at 23:05 -0700, Robert Goodyear wrote: Is anyone having issues with audio being passed inbound via Teliax? Trying to isolate an issue here. Thx, -Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using G729 in pass through mode
Obelix wrote: Is it possible to use G729 on asterisk without the license? Yes, as is clearly documented on the wiki :-) Will decoding the DTMF tones from the caller require G729 processing? No, because you cannot use inband DTMF with G.729 anyway. Since you will need to be using out-of-band DTMF (RFC2833 for SIP, or the only method on IAX2), then there is no issue with decoding the G.729 audio stream. Note that you will not be able to monitor/record calls without G.729 licenses, nor will you be able to play any audio files that are not already in G.729 format. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rxfax/txfax
Hi, I'm trying to send a fax from a Zap channel to another Zap channel and I can't - below the logs. How could I solve this? Thanks! In RECEIVER: -- Accepting call from '' to '1980' on channel 0/2, span 1 Urgent handler -- Executing Answer(Zap/2-1, ) in new stack Urgent handler Urgent handler -- Executing Goto(Zap/2-1, receive-fax|fax|1) in new stack Urgent handler -- Goto (receive-fax,fax,1) Urgent handler -- Executing RxFAX(Zap/2-1, /fax/rx.tif) in new stack Urgent handler DCS with final frame tag In state 9 Coarse carrier frequency 1699.90 (66) Training error 0.249420 Training succeeded (constellation mismatch 0.407855) In SENDER: -- Executing TxFAX(Zap/1-1, /fax/testX.tif|caller|debug) in new stack Urgent handler Slow carrier up Slow carrier down Slow carrier up DIS: 80 00 ce f4 80 80 81 80 80 80 18 DIS with final frame tag In state 10 DIS: Prefer 256 octet blocks Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm 2D coding Scan line length: 215mm Recording length: Unlimited Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm Minimum scan line time for higher resolutions: T15.4 = T7.7 North American Letter (215.9mm x 279.4mm) North American Legal (215.9mm x 355.6mm) DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 20ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Start sending document Start tx document Changed from phase 2 to 4 DCS: 83 00 c6 80 80 80 00 HDLC underflow in state 3 Changed from phase 4 to 6 Changed from phase 6 to 3 Slow carrier up Slow carrier down T4 timeout in state 4 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Remote SIP Connections
You can try to open up port for SIP 5060udp and RTP 10-2udp... (default setting) to your asterisk box. You will also have to specify that your extensions are nat=yes your externip=xxx.xxx.xxx.xxx (in SIP.conf) so that the SDP protocol will write the public IP and port translations for RTP (voice data). If this doesn't work, switch to IAX2 protocol- there are many hard-phones out there that support IAX2 protocol- You will only have to open up 4569udp on your firewall to your asterisk box and thats it. Better be careful with the RTP statement above as its not necessarily true for all implementations and configurations. If asterisk initiates the RTP negotiation, it will use udp source ports from the range shown above. However, each sip phone vendor (hard or soft) can choose whatever port range they want. XLite is in the 8,000 range Cisco 79x0's are in the 16384 to 32766 range etc. If a remote device initiates the RTP negotiation, it may not fall into the range that you've stated. (E.g., don't bank on your favorite itsp falling into that range.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to slow down dialing
I would like to know if it is possible to slow down the dialing process in asterisk. I have 4 of my 8 phone lines that are VoDSL. When we try and dial out these 4 VoDSL Lines, the number is often miss dialed, or incomplete. I added a wait before Asterisk tries to dial the whole number, but that has not solved my problem. If I use a regular phone and dial out these lines, they work fine. My assumption is that asterisk dial tones are too fast and I would like to try slowing them down, or spacing out each digit, to see if this helps. I am using two TDM04B cards to connect the 4 pots and 4 VoDSL Lines. Any help or ideas would be appreciated. Randy -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.9/42 - Release Date: 7/6/05 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cluecon, A mix of leading Open Source VoIP devlopers...
Just an FYI guys we have some of the leading open source developers and projects going to speak/showcase at Cluecon. These include: Mark Spencer - Asterisk Bob Andreasen - SIPFoundry Craig Southeren - OpenH323 David Sugar - Bayonne This should be an exciting event for all. Register Today! hope to see you there! Brian West Asterlink.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Logging SIP response codes
Is there a way to log SIP response codes without enabling verbose logging? Reason being is that from time to time I see a call fail on our primary provider and roll-over to our backup providers. If I happen to catch it on the console I can see the code 484 or similar. It would really help in troubleshooting with our primary provider if I could log those types of codes. Verbose just saves way to much stuff in the log files. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_rxfax does not receive
are you sharing IRQ on yuor zap device?what version of libtiff you have?be sure you not are sharing IRQ whit your zap and other devices andbe sure you have the more recent version of libtiff. On 7/6/05, Bohuslav Coufal [EMAIL PROTECTED] wrote: Hi all, I try to use app_rxfax. Aplication app_rxfax start O.K., fax trying to send, but it will stop at the beginning of page and after few seconds it stop with error 400. Does anybody has any suggestions? Thanks, Bob. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Romero## ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and wireless on site personal paging system
hi patrick, Patrick schrieb: Did you try contacting the vendor of the base stations to see if they have a EuroISDN firmware update? My Eicon Diva Server BRI card supports the 1TR6 protocol. The firmware can be found here: ftp://ftp.isdn4linux.org/pub/isdn4linux/utils/eicon/firmware/ Perhaps AVM supports 1TR6 too. yes, eicon diva server supports (we also have one here) but i was not able to load the capi drivers upon the 1TR6 stack?!? the next problem would be, that we need a isdn interface in NT mode, which is (to my knowledge) only possible with the cologne chip cards (junghanns / beronet). so i think we need an new solutions with the old wireless pagers. is there anybody who has experience with http://www.ascom.com/ws ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN PRI No Audio
On Wed, Jul 06, 2005 at 05:24:06PM -0500, Andy Brezinsky wrote: [Span 3 D-Channel 0] Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0 [Span 3 D-Channel 0]ChanSel: Reserved [Span 3 D-Channel 0] Ext: 1 DS1 Identifier: 2 [Span 3 D-Channel 0] Ext: 1 Coding: 0 Number Specified Channel Type: 3 [Span 3 D-Channel 0] Ext: 1 Channel: 24 ] [1e 02 81 83] Make sure that your span map is correctly done. It looks like the destination b channel is channel 24 on span 2. Make sure that you have your DS1s plugged in in the correct order and it's using the right DS1 for this. The channel that chan_zap picked for that was 48, so make sure also that they are not numbering the DS1 identifier beginning with 0. You might want to see if you need to adjust your spanmap and related config in zapata.conf for all of this. -- Matthew Fredrickson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] res_config_mysql.so in CVS asterisk-addons broken?
Hi! I would like to use the realtime extension of Asterisk and got the latest asterisk-addons from CVS. Upon compiling things, I got a couple of error messages from app_addon_mysql... is it me, or are the files in the CVS broken? Thanks, Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom phones - any advice
Oops. I forgot to add the recommendation to use the most current stable release of all firmware/boot loader/OS for the SNOM's as that can make a significant difference. Also, it may be better to see if you can purchase them from a vendor that will also support you, if possible. Sorry about that. RandyW Randy Williams wrote: Greetings, We are just finishing a roll-out of 25 of the SNOM 190s with a SNOM 220 w/sidecar. The only gotcha that I found is that the SNOM 190s use rfc2833 for a default dtfm mode and not inband which is the default for the asterisk server. I haven't ironed out the Mass deployment functionality yet, but will do so. So with a tftp server running you should be fine. Generally speaking, of course. RandyW Patrick Fortin wrote: Hi We are about to buy several Snom phones. Does anyone have warnings or advices against these phones ? Our finalists were Cisco, Polycom and Snom. We will be using only the SIP protocol. Thanks Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO hangup Problem.....
Hello, I am getting problem for delay call hang-up with the below scenario: PSTN User (calling Party)---PSTN Line FXO with Asterisk Box-SIP IP Phone (called party) I am using X100P card with my Asterisk-1.0.7 box. I am also using Zaptel-1.0.7 version. When PSTN user makes call to my PSTN line and after getting IVR, PSTN user dial my SIP IP Phone extension, as soon as PSTN user gets one ring back tone, PSTN user cut off the current call. But SIP IP Phone rings till its timeout. I would appreciate if anyone give me solution for the above case. Regards Nahid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: app_conference and AGI
In article [EMAIL PROTECTED], Jean-Hugues ROBERT [EMAIL PROTECTED] wrote: But... what is wrong with MEETME_AGI_BACKGROUND ? (besides the fact that unfortunately it does not work for SIP channels due to the mixing not being done in the zaptel driver but app_meetme itself, sort of, AFAIK). It's the other way round. The mixing is always done in the zaptel driver. For non-Zap channels, MeetMe creates a Zap pseudo channel, and in its main loop it copies frames of audio in both directions between the non-Zap channel and the associated pseudo channel. The Zaptel driver mixes the audio in the pseudo channels and the real Zaptel channels. When using MEETME_AGI_BACKGROUND, the main loop that does the pseudo channel copying is not invoked, so only the hardware channels get mixed in the Zaptel driver. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming 800-number over IAX - first few words are cut-off
Ok can you tell me if you get any errors on a short free call? :P You forgot to tell us what version of asterisk on both ends... wen can only guess at this point what the problem might be. /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jul 6, 2005, at 2:04 PM, Joseph wrote: I have an incoming 800-number over IAX from Teliax and I'm experiencing the large packet loss on connection. When a call comes in there is no ring tone and the first few words of the welcome message are cut off, regardless of the delay I set. Standard call (not 800-number) coming over IAX with the same provider works just fine only the tall free number. So it seems there are some packet loss only at the beginning, as the call quality sounds just fine, even when I compile something and CPU is at 99% use, there is no packet drop during conversation only on connection of tall free number. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_config_mysql.so in CVS asterisk-addons broken?
Christoph wrote: Hi! I would like to use the realtime extension of Asterisk and got the latest asterisk-addons from CVS. Upon compiling things, I got a couple of error messages from app_addon_mysql... is it me, or are the files in the CVS broken? Thanks, Christoph Please explain why your email subject referes to res_config_mysql but your email says absolutly nothing about it? The files in CVS are not broken. I'm using them right now in a prod environment. What errors are you getting? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_config_mysql.so in CVS asterisk-addons broken?
Hi, I spent quite a few days with this and in the end I find that the 1.07 release is by far the most stable. I had a lot of trouble with the CVS release. Ofcourse, thats just in my case, what do the others feel on this? Regards, Sahil Gupta VoiceValley On Thu, 7 Jul 2005, Christoph wrote: Hi! I would like to use the realtime extension of Asterisk and got the latest asterisk-addons from CVS. Upon compiling things, I got a couple of error messages from app_addon_mysql... is it me, or are the files in the CVS broken? Thanks, Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with iax2 and 2 peers behind nat
HI all, thanks Carlos, now its all working, but i have other cuestion, how y transfer call to other peer, when i try sip y do it pressing the # key but with iax it is not working. - Original Message - From: Carlos Alperin To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, July 06, 2005 7:06 PM Subject: RE: [Asterisk-Users] problem with iax2 and 2 peers behind nat Juan, That is not going to work. Asterisk shouldnt be behind a NAT to get registration of boxes behind NAT. Put the asterisk on DMZ zone of their router to make that happen. Carlos Alperin [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRIDSent: Wednesday, July 06, 2005 12:52 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] problem with iax2 and 2 peers behind nat Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan, itried with 2 firefly and the error its the same, it could be because the 2 peers are going to the internet with the same ip addres(both behind nat)? if i conect both peers in the same lan there is no problem so i think it cpuld be a problem with nat, i dont konw if i had to change some configuration in iax.conf. Thanks. Juan Lopez. [EMAIL PROTECTED] Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es _Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Transfers
I'm having a strange problem with transfers on IAX phones. I have two IAX phones behind my firewall that are extensions from my office phone system. Both phones can receive calls, but only one of the extensions can do blind transfers by pressing the # key. I have a similar problem at the office. Some of the phones can transfer calls, some of them can't. And my Zap lines can always transfer. I have all of my IAX extensions configured exactly the same way in iax.conf. All handsets are configured the same way and runnign the same firmware. I thought at first that it was a problem with NAT, but none of the office phones are behind firewalls. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXphone - ip address - extension number.
Hi, I'm trying to set up two ACT SIP/IAX capable phones to communicate with each other on the same internal network, using asterisk 1.0.9 on SuSE 9.3 (because I intend to grow the situation after this basic setup is functioning) The phone IPs are set to 192.168.0.201 and 202 respectively. I've had a look at iax.conf and extensions.conf but cannot see how to tie these IPs to an extension number, let alone how to dial that extension. The Getting Started with Asterisk and the Asterisk Doc Proj - Vol 1 that I have been using just have far too much info to work out what can be ignored in order to get such a simple setup working. I'd be happy for any help or pointers to steps that I should have followed. TIA, Zoltan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received
I have this problem zaphfc: empty HDLC frame or bad CRC received My configurations are cat /proc/zaptel/1 Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [NT] layer 1 ACTIVATED (G3) AMI/CCS 1 ZTHFC1/0/1 Clear 2 ZTHFC1/0/2 Clear 3 ZTHFC1/0/3 HDLCFCS cat /etc/zaptel.conf # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3 cat /etc/asterisk/zapata.conf ; ; Zapata telephony interface ; ; Configuration file [channels] language=it switchtype=euroisdn ; p2mp TE mode ;signalling=bri_cpe_ptmp ; p2p TE mode ;signalling=bri_cpe ; p2mp NT mode ;signalling=bri_net_ptmp ; p2p NT mode signalling=bri_net pridialplan=dynamic prilocaldialplan=local nationalprefix=0 internationalprefix=00 echocancel=yes echotraining=100 echocancelwhenbridged=yes immediate=yes group=1 context=default channel = 1 channel = 2 ztcfg -vv Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. Help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and wireless on site personal paging system
On Thu, 2005-07-07 at 17:04 +0200, Frank Sautter wrote:
hi patrick,
Patrick schrieb:
Did you try contacting the vendor of the base stations to see if they
have a EuroISDN firmware update? My Eicon Diva Server BRI card supports
the 1TR6 protocol. The firmware can be found here:
ftp://ftp.isdn4linux.org/pub/isdn4linux/utils/eicon/firmware/
Perhaps AVM supports 1TR6 too.
yes, eicon diva server supports (we also have one here) but i was not
able to load the capi drivers upon the 1TR6 stack?!?
Here is what I did to make it work with ETSI/TE on my box:
If you use Fedora Core or another distro that uses udev than you
probably have to add the following file in /etc/udev/rules.d *before*
modprobing the modules:
-- start file 10-capi.rules --
SYSFS{dev}=68:0, NAME=capi20
SYSFS{dev}=191:[0-9]*,NAME=capi/%n
-- end file 10-capi.rules --
Here is the order in which I load the kernel capi modules
from /etc/rc.d/rc.local:
# Start the Eicon card
/sbin/modprobe -v divas
sleep 5
/sbin/modprobe -v diva_idi
sleep 5
/sbin/modprobe -v kernelcapi
sleep 5
/sbin/modprobe capi
sleep 5
/sbin/modprobe divacapi
sleep 5
/sbin/divactrl load -c 1 -f ETSI -s 1 -vd6
sleep 5
The sleep 5 is needed to give udev some time to generate the proper
devices. I don't know exactly which module triggers it so I put a sleep
5 after each modprobe.
After you have manually activated the modules divactrl above,
check /var/log/messages for any erros and the correct activation of the
card with /usr/bin/capiinfo. My output is something like:
[EMAIL PROTECTED] ~]# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: Eicon Networks
CAPI Version: 2.0
Manufacturer Version: 2.0
Serial Number: 1884
BChannels: 2
[snip rest of output]
the next problem would be, that we need a isdn interface in NT mode,
which is (to my knowledge) only possible with the cologne chip cards
(junghanns / beronet).
Yes those cards support NT mode. Loading the 1TR6 protocol on the Eicon
card would be done by first putting the 1TR6 firmware files (see url
previously mentioned) in /urs/share/eicon (for divactrl-2.1) and then
do:
/sbin/divactrl load -c 1 -f 1TR6
For NT mode I think you need to specify -s 2 too although the help
output from /sbin/divactrl ctrl mentions PRI and not BRI.
Hope this helps.
Regards,
Patrick
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Re: [Asterisk-Users] Re: app_conference and AGI
At 15:31 07/07/2005 +, Tony Mountifield wrote: In article [EMAIL PROTECTED], Jean-Hugues ROBERT [EMAIL PROTECTED] wrote: But... what is wrong with MEETME_AGI_BACKGROUND ? (besides the fact that unfortunately it does not work for SIP channels due to the mixing not being done in the zaptel driver but app_meetme itself, sort of, AFAIK). It's the other way round. The mixing is always done in the zaptel driver. For non-Zap channels, MeetMe creates a Zap pseudo channel, and in its main loop it copies frames of audio in both directions between the non-Zap channel and the associated pseudo channel. The Zaptel driver mixes the audio in the pseudo channels and the real Zaptel channels. When using MEETME_AGI_BACKGROUND, the main loop that does the pseudo channel copying is not invoked, so only the hardware channels get mixed in the Zaptel driver. Cheers Tony Thanks for the clarification Tony. Any idea on how to make it so that MEETME_AGI_BACKGROUND would work on SIP channels (well, I suspect the issue is there with IAX too or any VoIP channel for that matter...) ? Maybe there could be a thread that would do what the main loop does. But... there might be an issue if the two threads (the one dealing with AGI and the main loop one) both try to read the frames... If this is not possible, then maybe the copying should occur earlier, before frame is delivered to the AGI ? This may require an additional data member in the channel structure. Well... this is kind of beyond my current needs/knowledge. OTOH, isn't recording done in a distinct thread ? If so, then the same kind of solution might be feasible. Thanks again for the clarification. Yours, JeanHuguesRobert - Web: http://hdl.handle.net/1030.37/1.1 Phone: +33 (0) 4 92 27 74 17 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Remote SIP Connections
Ok... You will need to give us more information... What type of SIP Phones are you using?? (Make and Model) What model of WRT54G are you using? What firmware do you have on the WRT54G? Regards, Derek - Original Message - From: Blake Krone [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 06, 2005 9:28 PM Subject: [Asterisk-Users] Re: Remote SIP Connections forgot to include the list -- Forwarded message -- From: Blake Krone [EMAIL PROTECTED] Date: Jul 6, 2005 9:07 PM Subject: Re: [Asterisk-Users] Re: Remote SIP Connections To: dbruce [EMAIL PROTECTED] Just had my brother connect from his time warner cable in minnesota to my adelphia in colorado springs, both NAT'd and I have my DMZ on, still nothing :( Any other ideas??? I wanted to setup an asterisk server so I could have VoIP in the house but then send SIP phones to my parents in Minnesota to save on long distance costs and cell minute usage. Thanks! On 7/5/05, Blake Krone [EMAIL PROTECTED] wrote: Well I had it setup with DMZ and port forwarding, removed the port forwards and still no luck :( Might end up going back to @home seen as other things like music on hold won't work properly, maybe something is just messed up in my gentoo install of asterisk. -Blake On 7/5/05, dbruce [EMAIL PROTECTED] wrote: You have forgotten that the WRT54G is a NAT router. The phones that are trying to connect to your server are also very likely to be behind a NAT router. This make it almost impossible to tell what ports are actually going to be used for inbound or outbound traffic... many NAT routers do not attach any significance to SIP protocol messages. Add to that the fact that many IP phones do not use the same port range for RTP that asterisk uses by default, and you have a VERY difficult time determining which port ranges need to be forwarded. Your easiest solution is to remove the forwarding rules, give your asterisk server a static IP address on your local network, and configure that IP address as the DMZ. All unsolicited requests to the router are sent to the IP address configured as the DMZ. The DMZ settings are found under the Applications Gaming tab on the WRT54G. You could also play with port triggering settings, but that is also a very dificult process. Regards, Derek Bruce - Original Message - From: Blake Krone [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, July 05, 2005 7:10 PM Subject: [Asterisk-Users] Re: Remote SIP Connections I have gotten them to be able to connect but I am unable to hear the other person and they can't hear me either. What else am I missing? On 7/5/05, Blake Krone [EMAIL PROTECTED] wrote: Hello all, I have my * server setup behind a Linksys WRT54G on Adelphia cable. I have forwarded 5060,1-10020, and another port set can't remember off the top of my head but I can't seem to connect to the * server from any locations that are direct connects to the Internet. Am I missing a portset for forwarding? If I use the name service (voip.*.com) from my home connection on the same LAN as the * server it will connect fine. Any ideas? TIA! -blake ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to slow down dialing
Randy MacKay wrote: I would like to know if it is possible to slow down the dialing process in asterisk. I have 4 of my 8 phone lines that are VoDSL. When we try and dial out these 4 VoDSL Lines, the number is often miss dialed, or incomplete. I added a wait before Asterisk tries to dial the whole number, but that has not solved my problem. If I use a regular phone and dial out these lines, they work fine. My assumption is that asterisk dial tones are too fast and I would like to try slowing them down, or spacing out each digit, to see if this helps. I am using two TDM04B cards to connect the 4 pots and 4 VoDSL Lines. Any help or ideas would be appreciated. Randy Assume you are dialing DTMF - The DURATION of Asterisk generated tones can be one source of the problem. Those smarter with the code can be more specific, but MANY telco related systems generate tones that are too short , 75-80 Ms should work, but frequently they are as short as 50 Ms. Interdigit time is another possibility. Both probably can be adjusted in the source and recompiled, but the smarter code guys need to address that. Inserting multiple w in the dial string will mask any slow dialtone issue, as Asterisk doesn't detect dialtone either. As an aside, it has been found that the DETECTION of dial pulses ( remember pulse dialing? ) inbound on a TDM FXS interface is also too restrictive, and can be corrected in the driver source. Anyone interested, E-mail me off list. John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_config_mysql.so in CVS asterisk-addons broken?
Sahil Gupta wrote: Hi, I spent quite a few days with this and in the end I find that the 1.07 release is by far the most stable. I had a lot of trouble with the CVS release. Ofcourse, thats just in my case, what do the others feel on this? Regards, Sahil Gupta VoiceValley Been using CVS-HEAD in production env with 80 SIP UA's, and Digium T1 card for several months now. No crashes. No problems. Love it. Use RealTime for SIP registration, Extensions and Voicemail with res_config_mysql. No problems here. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAXphone - ip address - extension number.
What about define those phones on the SIP.conf and use sip, instead of IAX. That protocol use be more used to communicate Asterisk servers more than phones. Regards, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: Thursday, July 07, 2005 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAXphone - ip address - extension number. Hi, I'm trying to set up two ACT SIP/IAX capable phones to communicate with each other on the same internal network, using asterisk 1.0.9 on SuSE 9.3 (because I intend to grow the situation after this basic setup is functioning) The phone IPs are set to 192.168.0.201 and 202 respectively. I've had a look at iax.conf and extensions.conf but cannot see how to tie these IPs to an extension number, let alone how to dial that extension. The Getting Started with Asterisk and the Asterisk Doc Proj - Vol 1 that I have been using just have far too much info to work out what can be ignored in order to get such a simple setup working. I'd be happy for any help or pointers to steps that I should have followed. TIA, Zoltan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] problem with iax2 and 2 peers behind nat
Do you have different dialplan for IAX SIP?, that shoudnt depend on the protocol used. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRID Sent: Thursday, July 07, 2005 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] problem with iax2 and 2 peers behind nat HI all, thanks Carlos, now its all working, but i have other cuestion, how y transfer call to other peer, when i try sip y do it pressing the # key but with iax it is not working. - Original Message - From: Carlos Alperin To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, July 06, 2005 7:06 PM Subject: RE: [Asterisk-Users] problem with iax2 and 2 peers behind nat Juan, That is not going to work. Asterisk shouldnt be behind a NAT to get registration of boxes behind NAT. Put the asterisk on DMZ zone of their router to make that happen. Carlos Alperin [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRID Sent: Wednesday, July 06, 2005 12:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] problem with iax2 and 2 peers behind nat Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan, itried with 2 firefly and the error its the same, it could be because the 2 peers are going to the internet with the same ip addres(both behind nat)? if i conect both peers in the same lan there is no problem so i think it cpuld be a problem with nat, i dont konw if i had to change some configuration in iax.conf. Thanks. Juan Lopez. [EMAIL PROTECTED] Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es _ Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] URGENT: hardware spesifications needed
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Frisch Sent: Wednesday, July 06, 2005 4:22 PM To: Jimmy Smith; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] URGENT: hardware spesifications needed Come on now children. Is this not a place to share knowledge? Well..., yes, and no. Information that isn't readily available elsewhere may legitimately be sought here. However, when the question is of the FAQ variety, and it is clear that the person asking it has not even attempted to find the information for himself, then rude replies are not out of line, IMO. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.10/43 - Release Date: 07/06/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with iax2 and 2 peers behind nat
Hi carlos, the dialplan its the same, i have only change the line dial[sip/peer] by dial[aix2/peer]. - Original Message - From: Carlos Alperin To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, July 07, 2005 6:51 PM Subject: RE: [Asterisk-Users] problem with iax2 and 2 peers behind nat Do you have different dialplan for IAX SIP?, that shoudnt depend on the protocol used. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRIDSent: Thursday, July 07, 2005 12:27 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] problem with iax2 and 2 peers behind nat HI all, thanks Carlos, now its all working, but i have other cuestion, how y transfer call to other peer, when i try sip y do it pressing the # key but with iax it is not working. - Original Message - From: Carlos Alperin To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, July 06, 2005 7:06 PM Subject: RE: [Asterisk-Users] problem with iax2 and 2 peers behind nat Juan, That is not going to work. Asterisk shouldnt be behind a NAT to get registration of boxes behind NAT. Put the asterisk on DMZ zone of their router to make that happen. Carlos Alperin [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRIDSent: Wednesday, July 06, 2005 12:52 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] problem with iax2 and 2 peers behind nat Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan, itried with 2 firefly and the error its the same, it could be because the 2 peers are going to the internet with the same ip addres(both behind nat)? if i conect both peers in the same lan there is no problem so i think it cpuld be a problem with nat, i dont konw if i had to change some configuration in iax.conf. Thanks. Juan Lopez. [EMAIL PROTECTED] Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es _Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es _Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie Question: Type of card
Hi, I am sorta a newbie to the asterisk community at least in the realm of hardware types. I was wondering, what type of card is used to allow asterisk, on a slackware installation to talk to a standard phone line so that asterisk can call out? Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXphone - ip address - extension number.
Carlos Alperin wrote: What about define those phones on the SIP.conf and use sip, instead of IAX. That protocol use be more used to communicate Asterisk servers more than phones. Regards, Carlos Alperin Ah - ok - I understood from the docs that IAX was better and, as the phone was capable of both, I've been trying to get it going via IAX. regards, Zoltan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Question: Type of card
On Thu, 7 Jul 2005 10:49:32 -0700 Dan Adams [EMAIL PROTECTED] wrote: Hi, I am sorta a newbie to the asterisk community at least in the realm of hardware types. I was wondering, what type of card is used to allow asterisk, on a slackware installation to talk to a standard phone line so that asterisk can call out? Dan The link below gives you great information on the card you need. Look espicially close to the box with all the writing in it just below the URL to www.asterisk.org that starts with For interconnection with digital and analog telephony equipment http://www.voip-info.org/wiki-Asterisk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Question: Type of card
Take a look here: http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TDM400P MARK. Dan Adams wrote: Hi, I am sorta a newbie to the asterisk community at least in the realm of hardware types. I was wondering, what type of card is used to allow asterisk, on a slackware installation to talk to a standard phone line so that asterisk can call out? Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDMoE bandwidth and load
Hello, We've just started using TDMoE(local T1s connecting between Asterisk servers in the same building over the LAN) to connect a few of our high-availability servers instead of using crossover T1 cables. The 3 servers we have connected to each other over TDMoE are running just fine and we have no audio quality issues or bandwidth issues, but I'm considering using TDMoE to connect 8 other servers to a main server and was wondering if a single ethernet interface on the Main server can handle the load of 8 dynamic spans connecting to it from other Asterisk servers. Does anyone have any experience with using TDMoE to run 8 virtual T1s on a single Ethernet port? Thanks, MATT--- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] aah and astcc
Darren: Thanks for your interest I would like that once you have been verified you can use aah dial plan so you can get all the reports for the astcc calls Thanks for your help Erick Weber - Original Message - From: Darren Wiebe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 06, 2005 8:26 PM Subject: Re: [Asterisk-Users] aah and astcc How exactly are you thinking. So that a certain aah extension points to it or so that once you have been verified you can call aah extensions? Darren Erick Weber V. wrote: Hello: Does anyone know how to incorporate astcc to aah so it will use amah extensions. Any help will be appreciate Thanks Erick W. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Braodvoice - UK Non Geographic Numbers
Since May 05 I have been unable to call any non-geographic number in the UK via Broadvoice. Thse are numbers such as the 0800 range (free to call) 087xx (local / national rate calls). Broadvoice support have been unhelpful, and can't say if there's any intention to fix this. A case has been upen since May 24 without any updates. Is anyone else having this problem? Has anyone else spoken to broadvoice about it? Did you get any further? Is there any indication it might be resolved? The last customer rep I spoke to recommended I close my account if I need to dial these numbers - I'd prefer to keep my phone number, but if all else fails... Russell. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Braodvoice - UK Non Geographic Numbers
Russell Horn wrote: Since May 05 I have been unable to call any non-geographic number in the UK via Broadvoice. Thse are numbers such as the 0800 range (free to call) 087xx (local / national rate calls). Broadvoice support have been unhelpful, and can't say if there's any intention to fix this. A case has been upen since May 24 without any updates. Is anyone else having this problem? Has anyone else spoken to broadvoice about it? Did you get any further? Is there any indication it might be resolved? The last customer rep I spoke to recommended I close my account if I need to dial these numbers - I'd prefer to keep my phone number, but if all else fails... Russell. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I lost a client because of this. BT will not allow premium numbers to be called from outside the UK. I even tried it from an ITSP in the Netherlands, and the call didn't go through :-( The ATT monopoly is gone. Hopefully, BT's time will come--the sooner the better. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie Question: Type of card
Here is what I use: http://www.digitnetworks.com/store/product_info.php?cPath=22products_id=28 I have used it with Slack, but now I am running it with FC4. -Original Message- From: Dan Adams [mailto:[EMAIL PROTECTED] Sent: Thursday, July 07, 2005 12:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Newbie Question: Type of card Hi, I am sorta a newbie to the asterisk community at least in the realm of hardware types. I was wondering, what type of card is used to allow asterisk, on a slackware installation to talk to a standard phone line so that asterisk can call out? Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - A.G. Edwards Sons' outgoing and incoming e-mails are electronically archived and subject to review and/or disclosure to someone other than the recipient. - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Transfers
what does asterisk says in the console when you try to transfer from the buggy phones?? asterisk -vvr On 7/7/05, Brent Davidson [EMAIL PROTECTED] wrote: I'm having a strange problem with transfers on IAX phones. I have two IAX phones behind my firewall that are extensions from my office phone system. Both phones can receive calls, but only one of the extensions can do blind transfers by pressing the # key. I have a similar problem at the office. Some of the phones can transfer calls, some of them can't. And my Zap lines can always transfer. I have all of my IAX extensions configured exactly the same way in iax.conf. All handsets are configured the same way and runnign the same firmware. I thought at first that it was a problem with NAT, but none of the office phones are behind firewalls. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Braodvoice - UK Non Geographic Numbers
Broadvoice could connect to non geographic numbers without difficulty until the fourth week of May 2005. I can call non-geographic numbers from my land line in the US, my mobile phone and from any calling card I have tried. This isn't an issue with BT but with broadvoice and those they contract to supply connections to the UK PSTN. On 7/7/05, Michael Welter [EMAIL PROTECTED] wrote: Russell Horn wrote: Since May 05 I have been unable to call any non-geographic number in the UK via Broadvoice. Thse are numbers such as the 0800 range (free to call) 087xx (local / national rate calls). Broadvoice support have been unhelpful, and can't say if there's any intention to fix this. A case has been upen since May 24 without any updates. Is anyone else having this problem? Has anyone else spoken to broadvoice about it? Did you get any further? Is there any indication it might be resolved? The last customer rep I spoke to recommended I close my account if I need to dial these numbers - I'd prefer to keep my phone number, but if all else fails... Russell. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I lost a client because of this. BT will not allow premium numbers to be called from outside the UK. I even tried it from an ITSP in the Netherlands, and the call didn't go through :-( The ATT monopoly is gone. Hopefully, BT's time will come--the sooner the better. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO hangup Problem.....
i have similar problem, but the sip phone just rings 1 or 2 more times, not until the timeout expires. what is your config in zapata.conf specifically callprogress an busydetect parameters can help best regards On 7/7/05, Nahid Hossain [EMAIL PROTECTED] wrote: Hello, I am getting problem for delay call hang-up with the below scenario: PSTN User (calling Party)---àPSTN Line à FXO with Asterisk Box-àSIP IP Phone (called party) I am using X100P card with my Asterisk-1.0.7 box. I am also using Zaptel-1.0.7 version. When PSTN user makes call to my PSTN line and after getting IVR, PSTN user dial my SIP IP Phone extension, as soon as PSTN user gets one ring back tone, PSTN user cut off the current call. But SIP IP Phone rings till its timeout. I would appreciate if anyone give me solution for the above case. Regards Nahid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Braodvoice - UK Non Geographic Numbers
its not you, its their false advertising that makes you think you can dial these (after all their rates page *still* claims they provide service and that its unlimited based on plan). There are threads on voxilla.com in the broadvoice forums, which have chat logs between me and the CTO nathan stratton, along with a slew of phone numbers of people at broadvoice (managers and such). Basically Nathan said they found a UK provider to terminate these calls and then nothing ever happened, he did give me a credit on my account becuase I was unable to call, something I am going to have to get again since I still havent been able to call and NCFA/LCFA/FREE were the reasons I choose broadvoice vs someone else. There is also an interesting thread there on the ownership structure of broadvoice, and how broadvoice is a registered trademark of broadcomm being used without permission by broadvoice.com. On Thu, 2005-07-07 at 14:38 -0400, Russell Horn wrote: Since May 05 I have been unable to call any non-geographic number in the UK via Broadvoice. Thse are numbers such as the 0800 range (free to call) 087xx (local / national rate calls). Broadvoice support have been unhelpful, and can't say if there's any intention to fix this. A case has been upen since May 24 without any updates. Is anyone else having this problem? Has anyone else spoken to broadvoice about it? Did you get any further? Is there any indication it might be resolved? The last customer rep I spoke to recommended I close my account if I need to dial these numbers - I'd prefer to keep my phone number, but if all else fails... -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] URGENT: hardware spesifications needed
Who are you to decide what Information can and cannot be legitimately be sought here:? Just curious. --Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rusty Shackleford Sent: Thursday, July 07, 2005 12:03 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] URGENT: hardware spesifications needed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Frisch Sent: Wednesday, July 06, 2005 4:22 PM To: Jimmy Smith; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] URGENT: hardware spesifications needed Come on now children. Is this not a place to share knowledge? Well..., yes, and no. Information that isn't readily available elsewhere may legitimately be sought here. However, when the question is of the FAQ variety, and it is clear that the person asking it has not even attempted to find the information for himself, then rude replies are not out of line, IMO. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.8.10/43 - Release Date: 07/06/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1
Does anyone have comment on this? I am getting: NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 on my asterisk box and it seems to be causing a poping sound in the phones, I am wondering if anyone can shed some light on this. I have scanned the archives and get possibilities ranging form motherboards, to pri, to loaded module problems. Can someone tell me the best way to start tracking this down? -- Thanks, Lance Grover ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] changing Nobody picked up in 30000 ms
how to edit the time 3 ms for ringing to 4 ms, i ve tried but i dindt know how,so please help me please. __ Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Braodvoice - UK Non Geographic Numbers
On Thursday 07 July 2005 19:55, Russell Horn wrote: I can call non-geographic numbers from my land line in the US, my mobile phone and from any calling card I have tried. This isn't an issue with BT but with broadvoice and those they contract to supply connections to the UK PSTN. nod If BroadVoice don't let you call national rate numbers, then use a second ITSP for those routes, or switch completely. They're cheap and nasty, but they do use IAX... http://www.call1899.co.uk/voip.php http://www.call1899.co.uk/voiprates.php No signup fee, no contract, cheap rates, lukewarm reliability :) Suck it and see - better than no service at all. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Long Distance
Hello Everyone, Pardon me if im sounding like a total idiot, but im new to this and have to ask. Numerous people have been telling me that I will be able to somehow do long distance calling for free when I roll out Asterisk.. and yet none of them can explain to me how exactly that will be. So.. I'll ask the community at large.. is this total BS or is there actually a way to reduce my long distance charges by rolling this out? I appreciate any info provided.. Thanks! -Don Brearley PS: I'm planning on deploying asterisk on a 300-phone line campus, and this is all in the planning stage at this point. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Braodvoice - UK Non Geographic Numbers
My take on this is that they are protecting themselves against fraud. Discounting the freefone numbers for a while, the national rate numbers are charged at variying rates and so how is a company to know just what they are gonna get charged. Gavin Hamill wrote: On Thursday 07 July 2005 19:55, Russell Horn wrote: I can call non-geographic numbers from my land line in the US, my mobile phone and from any calling card I have tried. This isn't an issue with BT but with broadvoice and those they contract to supply connections to the UK PSTN. nod If BroadVoice don't let you call national rate numbers, then use a second ITSP for those routes, or switch completely. They're cheap and nasty, but they do use IAX... http://www.call1899.co.uk/voip.php http://www.call1899.co.uk/voiprates.php No signup fee, no contract, cheap rates, lukewarm reliability :) Suck it and see - better than no service at all. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Crashes after update
After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from CVS, Asterisk crashes on startup with an apparent MySQL (res_config_register) error: # asterisk -vvvgc asterisk_startup_error1.log asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: un defined symbol: ast_cust_config_register The log is shown below. I've seen the posts from 1/25/05 and several more recent ones regarding this same issue or a similar one with the ast_cust_config_register being undefined, however reverting to that build of 1/24/05 does not solve the problem in my case. Is there another issue with mySQL that may cause this problem? I'm using SUSE 9.3 on an Athlon 64 with 64 bit release 2.6 of Linux. I've made sure that all the ODBC and MySQL modules for SUSE 9.3 are installed. I'm a rank noob with * and would appreciate any help. Thanks!!! Log Pasted below for more info: [0;37;40m [1;30;40m == [0;37;40mParsing '/etc/asterisk/asterisk.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/asterisk.conf': Found Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = [1;30;40m == [0;37;40mParsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader loading preload modules: [1;30;40m == [0;37;40mParsing '/etc/asterisk/modules.conf': Found [1;30;40m == [0;37;40mManager registered action Ping [1;30;40m == [0;37;40mManager registered action Events [1;30;40m == [0;37;40mManager registered action Logoff [1;30;40m == [0;37;40mManager registered action Hangup [1;30;40m == [0;37;40mManager registered action Status [1;30;40m == [0;37;40mManager registered action Setvar [1;30;40m == [0;37;40mManager registered action Getvar [1;30;40m == [0;37;40mManager registered action Redirect [1;30;40m == [0;37;40mManager registered action Originate [1;30;40m == [0;37;40mManager registered action Command [1;30;40m == [0;37;40mManager registered action ExtensionState [1;30;40m == [0;37;40mManager registered action AbsoluteTimeout [1;30;40m == [0;37;40mManager registered action MailboxStatus [1;30;40m == [0;37;40mManager registered action MailboxCount [1;30;40m == [0;37;40mManager registered action ListCommands [1;30;40m == [0;37;40mParsing '/etc/asterisk/manager.conf': Found [1;30;40m == [0;37;40mParsing '/etc/asterisk/cdr.conf': Not found (No such file or directory) Jul 6 21:32:24 [1;33;40mNOTICE[0;37;40m[8492]: [1;37;40mcdr.c[0;37;40m:[1;37;40m1162[0;37;40m [1;37;40mdo_reload[0;37;40m: CDR simple logging enabled. [1;30;40m == [0;37;40mParsing '/etc/asterisk/rtp.conf': Found [1;30;40m == [0;37;40mRTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [1;30;40m [0;37;40m[AbsoluteTimeout] [1;30;40m == [0;37;40mRegistered application '[1;36;40mAbsoluteTimeout[0;37;40m' [1;30;40m [0;37;40m[Answer] [1;30;40m == [0;37;40mRegistered application '[1;36;40mAnswer[0;37;40m' [1;30;40m [0;37;40m[BackGround] [1;30;40m == [0;37;40mRegistered application '[1;36;40mBackGround[0;37;40m' [1;30;40m [0;37;40m[Busy] [1;30;40m == [0;37;40mRegistered application '[1;36;40mBusy[0;37;40m' [1;30;40m [0;37;40m[Congestion] [1;30;40m == [0;37;40mRegistered application '[1;36;40mCongestion[0;37;40m' [1;30;40m [0;37;40m[DigitTimeout] [1;30;40m == [0;37;40mRegistered application '[1;36;40mDigitTimeout[0;37;40m' [1;30;40m [0;37;40m[Goto] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGoto[0;37;40m' [1;30;40m [0;37;40m[GotoIf] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGotoIf[0;37;40m' [1;30;40m [0;37;40m[GotoIfTime] [1;30;40m == [0;37;40mRegistered application '[1;36;40mGotoIfTime[0;37;40m' [1;30;40m [0;37;40m[ExecIfTime] [1;30;40m == [0;37;40mRegistered application '[1;36;40mExecIfTime[0;37;40m' [1;30;40m [0;37;40m[Hangup] [1;30;40m == [0;37;40mRegistered application '[1;36;40mHangup[0;37;40m' [1;30;40m [0;37;40m[NoOp] [1;30;40m == [0;37;40mRegistered application '[1;36;40mNoOp[0;37;40m' [1;30;40m [0;37;40m[Prefix] [1;30;40m == [0;37;40mRegistered application '[1;36;40mPrefix[0;37;40m' [1;30;40m [0;37;40m[Progress] [1;30;40m == [0;37;40mRegistered application '[1;36;40mProgress[0;37;40m' [1;30;40m [0;37;40m[ResetCDR] [1;30;40m == [0;37;40mRegistered application '[1;36;40mResetCDR[0;37;40m' [1;30;40m [0;37;40m[ResponseTimeout] [1;30;40m == [0;37;40mRegistered application '[1;36;40mResponseTimeout[0;37;40m' [1;30;40m
Re: [Asterisk-Users] Long Distance
I don't have all the answers. You should be able to save money on LD because you can (in my experience) pick up substantially better rates for voip termination than typical pstn LD. You can get plans from some providers that allow unlimited long distance but it is all a balancing act. Hope this helps a little, Darren Wiebe [EMAIL PROTECTED] Don Brearley wrote: Hello Everyone, Pardon me if im sounding like a total idiot, but im new to this and have to ask. Numerous people have been telling me that I will be able to somehow do long distance calling for free when I roll out Asterisk.. and yet none of them can explain to me how exactly that will be. So.. I'll ask the community at large.. is this total BS or is there actually a way to reduce my long distance charges by rolling this out? I appreciate any info provided.. Thanks! -Don Brearley PS: I'm planning on deploying asterisk on a 300-phone line campus, and this is all in the planning stage at this point. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Parial Hang with cvs-HEAD and queues/agentcallbacklogin
Title: Parial Hang with cvs-HEAD and queues/agentcallbacklogin Hi Last night I upgraded an asterisk install from cvs of early this year to current cvs head and all seemed to be working OK, but now Im having several problems which seem to be related to queues. First off queues dont work, theres no error message, the channel just seems to hang cli output as follows: -- Executing Answer(Local/[EMAIL PROTECTED],2, ) in new stack Jul 7 20:09:46 WARNING[27638]: channel.c:640 channel_find_locked: Avoided initial deadlock for '0x86e3948', 10 retries! -- Executing Playback(Local/[EMAIL PROTECTED],2, support-welcome) in new stack -- Local/[EMAIL PROTECTED],1 answered SIP/ed-1-fc54 -- Playing 'support-welcome' (language 'en') == Spawn extension (itg, 800, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2ZOMBIE' -- Executing Set(SIP/ed-1-fc54, CALLERID(name)=Support) in new stack -- Executing Queue(SIP/ed-1-fc54, support|t|||180) in new stack When I hang up the dialling phone there is no cli ouput and show channels shows the channel as still there: SIP/ed-1-93ce (macro-queueinbound s 4 ) Up Queue support|t|||180 Calling an agent produces the same result, and show agents on the CLI produces no output. Were using dynamic agents with agentcallbacklogin. Other calls seem to proceed OK, although it does seem to be rather slow for instance 4 gotos and a set callerid takes approx 6 seconds. This is a low load system using no more than 3-4% cpu normally and asterisk isnt using an abnormal amount of cpu or memory. Does anyone have any ideas whats causing this, or how to set about debugging it further? Many thanks Ed ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
