[asterisk-users] RTP Mixer
Hi Just an assumption. After packets reach Asterisk, it does the conversion into the required format and forwards it to Zaptel driver, which in turn combines and sends one RTP stream back to Asterisk. How can a client check about number of participants etc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk with festival facing problem!!!!!
hi List, i've been trying to get festival work on my 1.4.4 *box for the last 3days, i've used the tutorial on this page http://www.voip-info.org/wiki-Asterisk+Festival+installation with exactly the same line in my dialplan just to make a test now when i try to call( dial 555 ) from my softphone i get this message on festival server debugger: serverTue May 8 11:36:53 2007 : Festival server started on port 1314 client(1) Tue May 8 11:37:31 2007 : accepted from localhost.localdomain client(1) Tue May 8 11:37:31 2007 : disconnected then from my CLI there nothing after parsing '/etc/asterisk/festival.conf' : found and my softphone get connected and can stay so till i hang up without any sound did someone esperienced this situation??? any clue?? thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] isup-oli or ani2
Hello, I am using asterisk and a2billing. Can some one tell me how can I get callingani2 field in a2billing. That way I will be able to identify if the call is from a pay phone. My telco provider is sending me isup-oli in the the from field. Or if there is another way to get the information if the caller is calling from a payphone. Thank you, -Jai ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with SPA3102
Hello, i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with cdr. Well all I want is to receive incoming calls from pstn on specified sip account (suppose 8000), and to initiate outgoing calls from all my asterisk sip accounts through SPA3102 device. Someone can explain me what may i set on SPA and asterisk to do this thing. Thank you for your support. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bug in voicemail module of Asterisk 1.4.2?
Dear users, I think I may found a bug in the voicemail module of Asterisk 1.4.2! Outgoing email notifications should use a real existing domain (let's call it domain.real) instead of the local domain (domain.local) so that some mail servers won't reject the mails. That's why I've set the serveremail option in voicemail.conf to [EMAIL PROTECTED] Unfortunately Asterisk is always sending these mails with the sender [EMAIL PROTECTED] regardless of the serveremail option. I was able to at least change this behavior to [EMAIL PROTECTED] by changing the line 192.168.100.1 hostname.domain.local hostname in /etc/hosts to 192.168.100.1 hostname.domain.local but not any further. I don't think this is a bug of the MTA (exim4) because sending mails via mutt does work, the emails are sent by [EMAIL PROTECTED] then. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?
Sven Jacobs wrote: Dear users, I think I may found a bug in the voicemail module of Asterisk 1.4.2! Outgoing email notifications should use a real existing domain (let's call it domain.real) instead of the local domain (domain.local) so that some mail servers won't reject the mails. That's why I've set the serveremail option in voicemail.conf to [EMAIL PROTECTED] Unfortunately Asterisk is always sending these mails with the sender [EMAIL PROTECTED] regardless of the serveremail option. You fix that in your mail-server with aliasing and/or canonicalising. I think the Asterisk behaviour is correct. It is similar to receiving an email from cron or some other daemon. That is sent from [EMAIL PROTECTED], which is fine for your internal purposes, but if you send it out externally, you'll need to map it to a external address. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to record CDR in DB Oracle
On 7 May 2007, at 17:27, Florian Overkamp wrote: Hi Everton, Everton Goularth wrote: I had success to do my asterisk to record CDR in a databese MYSQL... Now, I need to do it to record CDR in Oracle... Does Anybody knows how to do this?? Every hints are welcome There is no native Oracle driver available to my knowledge, but if you can install an ODBC driver for Oracle, Asterisk will happily use that. If anyone gets this to work, especially against an oracle instance on a separate machine, I'd love to know how you did it. I spent a day or so failing to get it to work, then gave up and had a perl script written that regularly posts the new CDR records to oracle over http(s). Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?
Per Jessen wrote: Outgoing email notifications should use a real existing domain (let's call it domain.real) instead of the local domain (domain.local) so that some mail servers won't reject the mails. That's why I've set the serveremail option in voicemail.conf to [EMAIL PROTECTED] Unfortunately Asterisk is always sending these mails with the sender [EMAIL PROTECTED] regardless of the serveremail option. You fix that in your mail-server with aliasing and/or canonicalising. I think the Asterisk behaviour is correct. It is similar to receiving an email from cron or some other daemon. That is sent from [EMAIL PROTECTED], which is fine for your internal purposes, but if you send it out externally, you'll need to map it to a external address. But then again I don't understand the serveremail option. What is it for then? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Could two Asterisk servers connect through VPN
On 7 May 2007, at 19:51, Gordon Henderson wrote: On Mon, 7 May 2007, Tielin Xu wrote: Hi list: Has anyone done to set up two servers in different remote offices through VPN in order to get the VoIP communication? Yes it will work, but depending on your hardware you might be better off not using the VPN and just using an IAX trunk over the public Internet (unless you're really paranoid about someone listening in) Even if you are paranoid, you can still just use IAX, set 'encryption=yes' at both ends and IAX will encrypt the calls for you. There is a bandwidth overhead, but it is probably less that that of a VPN. Note, the calling/called numbers are still passed in the clear over encrypted IAX, so you are still vulnerable to traffic analysis. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?
Sven Jacobs wrote: You fix that in your mail-server with aliasing and/or canonicalising. I think the Asterisk behaviour is correct. It is similar to receiving an email from cron or some other daemon. That is sent from [EMAIL PROTECTED], which is fine for your internal purposes, but if you send it out externally, you'll need to map it to a external address. But then again I don't understand the serveremail option. What is it for then? As far as I can tell (but I'm on 1.4.1), the serveremail option only sets the From-address, not the envelope-address. The envelope will probably always be asterisk-user@hostname /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 39
I will be out of the office until Monday, May 14. Please contact OWD at 800-337-3839 and ask for Client Services if you need immediate assistance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?
As far as I can tell (but I'm on 1.4.1), the serveremail option only sets the From-address, not the envelope-address. The envelope will probably always be asterisk-user@hostname The From-address ist set by the fromstring option - which works btw - so you are wrong :) Unfortunately setting the From-address does not fix my problem. Also see http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel compile error
hi vi /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c this file and look for line that says 2.6.19 change it to 2.6.18 and save and compile arun On 5/7/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, May 04, 2007 at 01:55:20PM -0400, mail-lists wrote: I get the following error when trying to compile zaptel on CentOS 5 kernel 2.6.18-8.1.3.el5 CC [M] /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c: In function â /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c:171: error: â has no member named â make[3]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o] Error 1 make[2]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp] Error 2 make[1]: *** [_module_/root/asterisk-src/zaptel-1.2.17.1] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-8.1.3.el5-i686' make: *** [all] Error 2 I'm kind of at my wits end with this - been trying for several hours.. Please test the patch in http://bugs.digium.com/view.php?id=9006 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel compile error
On Tue, May 08, 2007 at 01:27:03PM +0400, Arun Kumar wrote: hi vi /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c this file and look for line that says 2.6.19 change it to 2.6.18 and save and compile I repeat again: please test the patch in http://bugs.digium.com/view.php?id=9006 so other centos5 users will not have to manually edit that file. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?
Sven Jacobs wrote: As far as I can tell (but I'm on 1.4.1), the serveremail option only sets the From-address, not the envelope-address. The envelope will probably always be asterisk-user@hostname The From-address ist set by the fromstring option - which works btw - so you are wrong :) Unfortunately setting the From-address does not fix my problem. Maybe I'm misinterpreting things, but this is what I se: fromstring = the From:-text, not the From:-address. I'm just using the default fromstring, but I've set serveremail = asterisk@realdomain With this I get From: Asterisk PBX [EMAIL PROTECTED] Still, the envelope is [EMAIL PROTECTED]. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Beronet card - issue?
Hi all, I have a problem with my beronet card with 2 isdn. I think drivers and Asterisk are ok but the red led on the card always blinking. The card is connected with PBX. I post some conf: [EMAIL PROTECTED] ~]# misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX (maybe there is already a PBX running?) Port 2: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - childcnt: 2 mISDN_close: fid(3) isize(131072) inbuf(0x9eff060) irp(0x9eff060) iend(0x9eff060) [EMAIL PROTECTED] ~]# [EMAIL PROTECTED] ~]# cat /etc/misdn-init.conf card=1,0x4 te_ptmp=1,2 poll=127 dsp_poll=128 dsp_options=0 dtmfthreshold=100 debug=5 [EMAIL PROTECTED] ~]# cat /etc/asterisk/misdn.conf [general] debug = 0 method=standard bridging=no stop_tone_after_first_digit=yes append_digits2exten=yes dynamic_crypt=no crypt_prefix=** crypt_keys=test,muh [default] context=misdn language=it musicclass=default senddtmf=yes far_alerting=no allowed_bearers=all nationalprefix=0 internationalprefix=00 rxgain=0 txgain=0 te_choose_channel=no pmp_l1_check=no reject_cause=16 need_more_infos=no nttimeout=no method=standard dialplan=0 localdialplan=0 cpndialplan=0 early_bconnect=yes incoming_early_audio=no nodialtone=no callgroup=1 pickupgroup=1 presentation=-1 screen=-1 echocancel=yes jitterbuffer=4000 jitterbuffer_upper_threshold=0 hdlc=no [isdn] ports=1 context=from-pstn msns=* This is the first time that I configure this type of card Link of some good docs is ok too. :) Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk to record CDR in DB Oracle
Hi Tim, You will need an Oracle ODBC driver that Asterisk can use to connect to an oracle instance (local/remote). As far as I am aware, Oracle don't have unix/linux ODBC driver as of yet, but you can get one from EasySoft. They have an eval version you can try out to see if it works, have a look here: http://www.easysoft.com/products/data_access/odbc_oracle_driver/index.html I have tested this in the past and have managed to get asterisk to connect to a remote oracle instance using this driver, so it does work :) Thanks Bruce -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: 08 May 2007 09:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk to record CDR in DB Oracle On 7 May 2007, at 17:27, Florian Overkamp wrote: Hi Everton, Everton Goularth wrote: I had success to do my asterisk to record CDR in a databese MYSQL... Now, I need to do it to record CDR in Oracle... Does Anybody knows how to do this?? Every hints are welcome There is no native Oracle driver available to my knowledge, but if you can install an ODBC driver for Oracle, Asterisk will happily use that. If anyone gets this to work, especially against an oracle instance on a separate machine, I'd love to know how you did it. I spent a day or so failing to get it to work, then gave up and had a perl script written that regularly posts the new CDR records to oracle over http(s). Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.4.4 and Custom Postgres 8.2.4 (checkingfor PQexec in -lpq... no)
Hi Gavin, I don't know if this will help, but can you check to see if you have libtool installed? I had a similar issue with unixodbc, and once I installed libtool, it rectified the issue. Once libtool is installed, re-run configure and it should hopefully work. Thanks Bruce -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Henry Sent: 05 May 2007 22:31 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4.4 and Custom Postgres 8.2.4 (checkingfor PQexec in -lpq... no) Dear All, Why does my configure fail like so: checking for pg_config... /usr/local/pgsql/8.2.4/bin/pg_config checking for PQexec in -lpq... no configure: *** configure: *** The PostgreSQL installation on this system appears to be broken. configure: *** Either correct the installation, or run configure configure: *** including --without-postgres Configure options are: env CC=/usr/local/bin/gcc ./configure --with-ssl=/usr/local/ssl --with-postgres=/usr/local/pgsql/8.2.4 configure has found pg_config, what more does it need? I even tried: env CC=/usr/local/bin/gcc CPPFLAGS=-I/usr/local/pgsql/8.2.4/include \ LDFLAGS=-L/usr/local/pgsql/8.2.4/lib \ LD_LIBRARY_PATH=/usr/local/pgsql/8.2.4/lib ./configure --with-ssl=/usr/local/ssl --with-postgres=/usr/local/pgsql/8.2.4 Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax to iax Reject Connection
Hi, Don't you have to configure the host option for each channel in iax.conf? Look at: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf Ronaldo. chawki hammoud wrote: Hi: It's the first time I have this problem. No matter how I configure my two IAX machines the connection is rejected. chan_iax2.c:5550 socket_read: Call rejected by : No authority found iax server A: [saad_out] type=peer host=hostip username=username secret=secret disallow=all allow=gsm iax server B: [guest] type=user username=username secret=secret context=tele disallow=all allow=gsm Any suggestions of why the connection is refused. I have no firewall. Thanks We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. http://tv.yahoo.com/collections/265 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outgoing calls
hello friends, I have a problem when I call to outside (9) from IPs Telephones. the incomning calls are OK. in the console when I put sip debug peer 101 I have this lines: *CLI sip debug peer 101 SIP Debugging Enabled for IP: 10.0.0.9:5060 *CLI -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (13 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5a68d228 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '101' -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK max-forwards: 70 Content-Length: 0 --- (8 headers 0 lines) --- -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Found user '101' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Peer audio RTP is at port 10.0.0.9:10010 Found description format G729 Found description format G723 Found description format G723high Found description format PCMA Found description format PCMU Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 943833473 in SOME (domain 101) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' 12 headers, 0 lines Reliably Transmitting (no NAT) to 10.0.0.9:5060: OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK65c2f229;rport From: asterisk sip:[EMAIL PROTECTED];tag=as0cc11f28 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED]
Re: [asterisk-users] Queue Status
Hi, you can use an AGI to connect to asterisk manager and retrieve the info you need about the queue. Hope it helps Arun Kumar ha scritto: Hi I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Foà 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outgoing calls
hello friends, I have a problem when I call to outside (9) from IPs Telephones. the incomning calls are OK. in the console when I put sip debug peer 101 I have this lines: *CLI sip debug peer 101 SIP Debugging Enabled for IP: 10.0.0.9:5060 *CLI -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (13 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5a68d228 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '101' -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK max-forwards: 70 Content-Length: 0 --- (8 headers 0 lines) --- -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Found user '101' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Peer audio RTP is at port 10.0.0.9:10010 Found description format G729 Found description format G723 Found description format G723high Found description format PCMA Found description format PCMU Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 943833473 in SOME (domain 101) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' 12 headers, 0 lines Reliably Transmitting (no NAT) to 10.0.0.9:5060: OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK65c2f229;rport From: asterisk sip:[EMAIL PROTECTED];tag=as0cc11f28 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED]
[asterisk-users] outgoing calls
hello friends, I have a problem when I call to outside (9) from IPs Telephones. the incomning calls are OK. in the console when I put sip debug peer 101 I have this lines: *CLI sip debug peer 101 SIP Debugging Enabled for IP: 10.0.0.9:5060 *CLI -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (13 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5a68d228 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '101' -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK max-forwards: 70 Content-Length: 0 --- (8 headers 0 lines) --- -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Found user '101' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Peer audio RTP is at port 10.0.0.9:10010 Found description format G729 Found description format G723 Found description format G723high Found description format PCMA Found description format PCMU Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 943833473 in SOME (domain 101) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' 12 headers, 0 lines Reliably Transmitting (no NAT) to 10.0.0.9:5060: OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK65c2f229;rport From: asterisk sip:[EMAIL PROTECTED];tag=as0cc11f28 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED]
RE: [asterisk-users] outgoing calls
Hello Josu, In you're sip.conf you have the 2 phones configured that they are in the SOME context. Looking at the SOME contect in extensions.conf you only have the 2 phones defined. If you want to call ouside from the SOME context as well, you need to include the outgoing context there as well. Regards, Roelof Dijkstra Network Engineer EMEA Compuware Europe BV -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Josu Lazkano Lete Sent: Tuesday, May 08, 2007 1:36 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] outgoing calls hello friends, I have a problem when I call to outside (9) from IPs Telephones. the incomning calls are OK. in the console when I put sip debug peer 101 I have this lines: *CLI sip debug peer 101 SIP Debugging Enabled for IP: 10.0.0.9:5060 *CLI -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (13 headers 13 lines) --- Using INVITE request as basis request - mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5a68d228 Content-Length: 0 --- Scheduling destruction of call mailto:'[EMAIL PROTECTED]' '[EMAIL PROTECTED]' in 15000 ms Found user '101' -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 1 ACK max-forwards: 70 Content-Length: 0 --- (8 headers 0 lines) --- -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Found user '101' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Peer audio RTP is at port 10.0.0.9:10010 Found description format G729 Found description format G723 Found description format G723high Found description format PCMA Found description format PCMU Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 943833473 in SOME (domain 101) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- -- SIP read from 10.0.0.9:5060: ACK
[asterisk-users] Responding to SIP OPTIONS
I understand that it is customary for SIP User Agents to send OPTIONS packets every now and then to check that a peer is still alive and well. Indeed I understand that Asterisk itself sends them if qualify is set to yes in the peer configuration. How is one supposed to configure the dialplan so that Asterisk responds correctly to these requests? At the moment, I'm seeing Looking for s in default and then a 404 Not Found being returned - which can't be right. Thanks! Alex Lake DIGITAL MAIL LIMITED ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.2 from svn ... lock on shutdown
Hi, I hope this gets picked up by some bug marshall ... I have downloaded (yesterday) the 1.2 branch from svn ... When running: asterisk -c loaded modules: [modules] autoload=no load = pbx_functions.so load = pbx_config.so load = codec_a_mu.so load = format_pcm_alaw.so load = codec_ulaw.so load = codec_alaw.so load = format_pcm.so load = func_uri.so ;required by app_dial and chan_sip load = res_features.so load = app_dial.so ;playback and echo apps ... load = app_playback.so load = app_echo.so load = codec_gsm.so load = format_gsm.so load = format_wav_gsm.so load = chan_h323.so load = chan_sip.so load = chan_local.so When I do: stop now asterisk hangs up, but locks: *CLI stop now Beginning asterisk shutdown Executing last minute cleanups Asterisk cleanly ending (0). I attached gdb to the locked process: 0xb725af28 in std::_Rb_tree_increment () from /usr/lib/libstdc++.so.6 (gdb) bt #0 0xb725af28 in std::_Rb_tree_increment () from /usr/lib/libstdc++.so.6 #1 0xb793f304 in std::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::operator++ () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #2 0xb79881a0 in std::__distancestd::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #3 0xb79881cb in std::distancestd::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #4 0xb7989ee6 in std::_Rb_treePString, std::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase*, std::_Select1ststd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* , std::lessPString, std::allocatorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::erase () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #5 0xb7989f20 in std::mapPString, PFactoryOpalMediaFormat, PString::WorkerBase*, std::lessPString, std::allocatorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::erase () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #6 0xb7989f5a in PFactoryOpalMediaFormat, PString::Unregister_Internal () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #7 0xb7989f9d in PFactoryOpalMediaFormat, PString::Unregister () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #8 0xb7989fc9 in OpalPluginMediaFormat::~OpalPluginMediaFormat () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #9 0xb748bea1 in PAbstractList::RemoveAt () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #10 0xb74892e1 in PCollection::RemoveAll () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #11 0xb7489e25 in PAbstractList::DestroyContents () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #12 0xb7490152 in PContainer::Destruct () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #13 0xb791ca57 in PAbstractList::~PAbstractList () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #14 0xb79755c9 in PListOpalMediaFormat::~PList () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #15 0xb79828e7 in H323PluginCodecManager::~H323PluginCodecManager () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #16 0xb7e0d4f0 in exit () from /lib/tls/libc.so.6 #17 0x080bddd7 in quit_handler (num=135324439, nice=3, safeshutdown=1, restart=0) at asterisk.c:945 #18 0x080be019 in handle_shutdown_now (fd=1, argc=2, argv=0xb830) at asterisk.c:1104 #19 0x0809811b in ast_cli_command (fd=1, s=0x8151900 \001) at cli.c:1364 #20 0x080c0d93 in main (argc=2, argv=0xbd84) at asterisk.c:1019 (gdb) Regards, Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?
Maybe I'm misinterpreting things, but this is what I se: fromstring = the From:-text, not the From:-address. I'm just using the default fromstring, but I've set serveremail = asterisk@realdomain With this I get From: Asterisk PBX [EMAIL PROTECTED] Still, the envelope is [EMAIL PROTECTED]. Taking your example I would get From: Asterisk PBX [EMAIL PROTECTED] Envelope: [EMAIL PROTECTED] so I guess there's something wrong here... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Status
Hi I already tried asterisk manager but Im not able to get status for each queue member. thanks On 5/8/07, Edoardo Serra [EMAIL PROTECTED] wrote: Hi, you can use an AGI to connect to asterisk manager and retrieve the info you need about the queue. Hope it helps Arun Kumar ha scritto: Hi I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Foà 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outgoing calls
thank you very much! it works - Original Message - From: Dijkstra, Roelof To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, May 08, 2007 1:53 PM Subject: RE: [asterisk-users] outgoing calls Hello Josu, In you're sip.conf you have the 2 phones configured that they are in the SOME context. Looking at the SOME contect in extensions.conf you only have the 2 phones defined. If you want to call ouside from the SOME context as well, you need to include the outgoing context there as well. Regards, Roelof Dijkstra Network Engineer EMEA Compuware Europe BV -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Josu Lazkano Lete Sent: Tuesday, May 08, 2007 1:36 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] outgoing calls hello friends, I have a problem when I call to outside (9) from IPs Telephones. the incomning calls are OK. in the console when I put sip debug peer 101 I have this lines: *CLI sip debug peer 101 SIP Debugging Enabled for IP: 10.0.0.9:5060 *CLI -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (13 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Reliably Transmitting (no NAT) to 10.0.0.9:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;received=10.0.0.9;rport=5060 From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5a68d228 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '101' -- SIP read from 10.0.0.9:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED];tag=as705b7b72 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK max-forwards: 70 Content-Length: 0 --- (8 headers 0 lines) --- -- SIP read from 10.0.0.9:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport From: 101 sip:[EMAIL PROTECTED];tag=3159122210 To: 943833473 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=101, realm=asterisk, nonce=5a68d228, uri=sip:[EMAIL PROTECTED], response=0fa2c9d246b54d137046e23743e1951c, algorithm=MD5 max-forwards: 70 supported: 100rel user-agent: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 278 v=0 o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9 s=A conversation c=IN IP4 10.0.0.9 t=0 0 m=audio 10010 RTP/AVP 18 4 4 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 10.0.0.9 : 5060 (NAT) Found user '101' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Peer audio RTP is at port 10.0.0.9:10010 Found description format G729 Found description format G723 Found description format G723high Found description format PCMA Found description format PCMU Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 943833473 in SOME
Re: [asterisk-users] Asterisk 1.4.4 and Custom Postgres 8.2.4 (checkingfor PQexec in -lpq... no)
On 08/05/07, Bruce McAlister [EMAIL PROTECTED] wrote: Hi Gavin, I don't know if this will help, but can you check to see if you have libtool installed? I had a similar issue with unixodbc, and once I installed libtool, it rectified the issue. Once libtool is installed, re-run configure and it should hopefully work. Digging in config.log revealed that I hadn't run ldconfig and linked some other custom libs correctly. All working now and contributed back missing files/docs etc.: http://bugs.digium.com/view.php?id=9676 Gavin Thanks Bruce -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Henry Sent: 05 May 2007 22:31 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4.4 and Custom Postgres 8.2.4 (checkingfor PQexec in -lpq... no) Dear All, Why does my configure fail like so: checking for pg_config... /usr/local/pgsql/8.2.4/bin/pg_config checking for PQexec in -lpq... no configure: *** configure: *** The PostgreSQL installation on this system appears to be broken. configure: *** Either correct the installation, or run configure configure: *** including --without-postgres Configure options are: env CC=/usr/local/bin/gcc ./configure --with-ssl=/usr/local/ssl --with-postgres=/usr/local/pgsql/8.2.4 configure has found pg_config, what more does it need? I even tried: env CC=/usr/local/bin/gcc CPPFLAGS=-I/usr/local/pgsql/8.2.4/include \ LDFLAGS=-L/usr/local/pgsql/8.2.4/lib \ LD_LIBRARY_PATH=/usr/local/pgsql/8.2.4/lib ./configure --with-ssl=/usr/local/ssl --with-postgres=/usr/local/pgsql/8.2.4 Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma A101 on Freebsd 6.2
I have an issue that hopefully you can help me solve. I've got the sangoma a101 card and installed it on freebsd but I according to the manual I should be see when running dmesg PCi0 vendor. something that tells me sangoma it's being recognized by the system. Now this is the 2nd card I try, the first one according to support was faulty and they sent me a new one. Is it possible to be doing something wrong? Is freebsd maybe recognizing differently from what it says on the manual? Please any help would be highly appreciated. -Zvonimir___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] load modules
Hello again, I have a little problem, every time I switch on the Asterisk server I must load two modules: modprobe zaptel and modprobe wctdm Is there any way to load there automatically when the server start? I have a Debian Etch. One more cuestion, it's posible to start Asterisk (asterisk -vvvc)as well? What metod do you prefer? asterisk or asterisk -vvvc? Thanks very much to all of you. Bye.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Send SIP Re-invite.
Rohan Hathiwala wrote: Hi, I need asterisk to instruct the other side to send RTP to a conference server running on a different machine. The conference server does not understand SIP so I cannot use the SIP REFER method. I have another question. Suppose when processing a SIP INVITE we want to use asterisk only for call control and let another server handle the RTP is there a clean way to do this in asterisk. Regards, Rohan Hathiwala. Asterisk/chan_sip wasn't designed to be able to do this. You're going to end up modifying things... potentially a lot. If the conference server does SIP though you can just dial it, make sure canreinvite is set to yes, and audio should go direct. Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?
Sven Jacobs wrote: Maybe I'm misinterpreting things, but this is what I se: fromstring = the From:-text, not the From:-address. I'm just using the default fromstring, but I've set serveremail = asterisk@realdomain With this I get From: Asterisk PBX [EMAIL PROTECTED] Still, the envelope is [EMAIL PROTECTED]. Taking your example I would get From: Asterisk PBX [EMAIL PROTECTED] Envelope: [EMAIL PROTECTED] so I guess there's something wrong here... The voicemail email gets handed off to sendmail for actual sending. It's adding on the envelope above. Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2 from svn ... lock on shutdown
Cesc wrote: Hi, I hope this gets picked up by some bug marshall ... Eep! Filing a bug is best instead of email it here for future reference... I attached gdb to the locked process: 0xb725af28 in std::_Rb_tree_increment () from /usr/lib/libstdc++.so.6 (gdb) bt #0 0xb725af28 in std::_Rb_tree_increment () from /usr/lib/libstdc++.so.6 #1 0xb793f304 in std::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::operator++ () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #2 0xb79881a0 in std::__distancestd::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #3 0xb79881cb in std::distancestd::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #4 0xb7989ee6 in std::_Rb_treePString, std::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase*, std::_Select1ststd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* , std::lessPString, std::allocatorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::erase () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #5 0xb7989f20 in std::mapPString, PFactoryOpalMediaFormat, PString::WorkerBase*, std::lessPString, std::allocatorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::erase () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #6 0xb7989f5a in PFactoryOpalMediaFormat, PString::Unregister_Internal () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #7 0xb7989f9d in PFactoryOpalMediaFormat, PString::Unregister () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #8 0xb7989fc9 in OpalPluginMediaFormat::~OpalPluginMediaFormat () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #9 0xb748bea1 in PAbstractList::RemoveAt () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #10 0xb74892e1 in PCollection::RemoveAll () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #11 0xb7489e25 in PAbstractList::DestroyContents () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #12 0xb7490152 in PContainer::Destruct () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #13 0xb791ca57 in PAbstractList::~PAbstractList () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #14 0xb79755c9 in PListOpalMediaFormat::~PList () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #15 0xb79828e7 in H323PluginCodecManager::~H323PluginCodecManager () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #16 0xb7e0d4f0 in exit () from /lib/tls/libc.so.6 #17 0x080bddd7 in quit_handler (num=135324439, nice=3, safeshutdown=1, restart=0) at asterisk.c:945 #18 0x080be019 in handle_shutdown_now (fd=1, argc=2, argv=0xb830) at asterisk.c:1104 #19 0x0809811b in ast_cli_command (fd=1, s=0x8151900 \001) at cli.c:1364 #20 0x080c0d93 in main (argc=2, argv=0xbd84) at asterisk.c:1019 (gdb) This is definitely an issue with chan_h323 and OpenH323. If you don't load chan_h323 can you then shut down fine? If so please file a bug on bugs.digium.com and the individual who looks after that stuff will look at it. Thanks! Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] isup-oli or ani2
JK wrote: Hello, I am using asterisk and a2billing. Can some one tell me how can I get callingani2 field in a2billing. That way I will be able to identify if the call is from a pay phone. My telco provider is sending me isup-oli in the the from field. Or if there is another way to get the information if the caller is calling from a payphone. Thank you, -Jai It sounds like your call is being delivered over SIP and they are sending the information that way. If so support is not built in to chan_sip to get that information, but you may be able to use the SIP_HEADER dialplan function to get that specific header and then CUT to get the specific part you need. Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax to iax Reject Connection
chawki hammoud wrote:
Hi:
It's the first time I have this problem.
No matter how I configure my two IAX machines the
connection is rejected.
chan_iax2.c:5550 socket_read: Call rejected by :
No authority found
Without seeing your dialplan it is a little hard to determine why but
I'll try based on your iax.conf entries below.
iax server A:
[saad_out]
type=peer
host=hostip
username=username
secret=secret
disallow=all
allow=gsm
When the call is sent out it is going to authenticate using the username
username.
iax server B:
[guest]
type=user
username=username
secret=secret
context=tele
disallow=all
allow=gsm
This is incorrect, username is not valid in a user entry. The username
is specified between the [ and ]. In this case it is guest. Therefore no
user entry exists with the username username and the other box can't
authenticate.
Any suggestions of why the connection is refused. I
have no firewall.
Thanks
Here is a basic entry for a peer:
[trunk-out]
type=peer
host=my.silly.box
username=myserver
secret=password
disallow=all
allow=ulaw
The call will be sent to my.silly.box and will try to authenticate as
myserver with the password password.
Here is a basic entry for a user:
[myserver]
type=user
secret=password
disallow=all
allow=ulaw
context=servers
Here is the respective dial line:
IAX2/trunk-out/${EXTEN}
Joshua Colp
Software Developer
Digium, Inc.
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Re: [asterisk-users] Sangoma A101 on Freebsd 6.2
Hi Zvonimir I have an issue that hopefully you can help me solve. I've got the sangoma a101 card and installed it on freebsd but I according to the manual I should be see when running dmesg PCi0 vendor. something that tells me sangoma it's being recognized by the system. Now this is the 2nd card I try, the first one according to support was faulty and they sent me a new one. Is it possible to be doing something wrong? Is freebsd maybe recognizing differently from what it says on the manual? This is probably not the best place to ask a configuration question about a Songoma card. Can Sangoma support help you? Have you tried the card in another system? - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: h323 problem with asterisk 1.2.18
On 5/7/07, nik600 [EMAIL PROTECTED] wrote: i am experiencing problem with asterisk 1.2.18 I've downloaded and installed pwlib and openh323 with the following commands: cd /path/to/pwlib ./configure make clean opt cd /path/to/openh323 ./configure make clean opt then 'ive set the corresponding PATH PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/ export PWLIBDIR OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/ export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH but when i go to: cd asterisk-1.2.18/channels/h323/ and do a make opt: [EMAIL PROTECTED]:/data/programmi/asterisk_1.2.18/asterisk-1.2.18/channels/h323# make opt make: *** No rule to make target `opt'. Stop. why? where am i wrong? i've also tried the last version of pwlib and openh323, but without fixing the problem thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser i've also tried supported version Open H.323 version v1.17.1, PWLib v1.9.0 but.. it doesn't compile. It seems to be a problem with makefile -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Responding to SIP OPTIONS
Alex Lake wrote: I understand that it is customary for SIP User Agents to send OPTIONS packets every now and then to check that a peer is still alive and well. Indeed I understand that Asterisk itself sends them if qualify is set to yes in the peer configuration. How is one supposed to configure the dialplan so that Asterisk responds correctly to these requests? At the moment, I'm seeing Looking for s in default and then a 404 Not Found being returned - which can't be right. Thanks! Alex Lake DIGITAL MAIL LIMITED Handling of OPTIONS in Asterisk has changed a little bit through chan_sip versions... but for the most part the other side usually just wants you to respond with something/anything. Is the other side unhappy with the 404 Not Found? Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A101 on Freebsd 6.2
Zvonimir Mileta wrote: I have an issue that hopefully you can help me solve. I've got the sangoma a101 card and installed it on freebsd but I according to the manual I should be see when running dmesg PCi0 vendor. something that tells me sangoma it's being recognized by the system. Now this is the 2nd card I try, the first one according to support was faulty and they sent me a new one. Is it possible to be doing something wrong? Is freebsd maybe recognizing differently from what it says on the manual? Please any help would be highly appreciated. -Zvonimir I would suggest talking to Sangoma support. They are extremely helpful and should be able to answer your questions in no time, give them a ring. Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GXV-3000 IP Video Phone
Hi Nitesh - Thanks everyone... The GXV-3000 IP Video Phone works with Asterisk 1.2 using H.263 Video Coder. I had to update both phones firmware with new one... Out of curiosity - do you like the phone? I've looked for reviews, but I haven't found any that rate the phone's functionality. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote Phone and Server Behind NAT
I have an asterisk Server (2.1.17) behind NAT with a static IP and port forwarding enabled. The remote SIP phone is also behind NAT. I've gotten them to work except when I specify a secret.When there is a secret configured the phone can not authorize. Has anyone gotten this to work? Regards, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 39
Stewart, till what time on Monday will you be out? On 8 May 2007 04:10:44 -0500, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I will be out of the office until Monday, May 14. Please contact OWD at 800-337-3839 and ask for Client Services if you need immediate assistance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outbound call through a Single Asterisk Server
I have two asterisk servers. One is at location 1 and the other is at
location 2.
What I am trying to do seems straightforward. I want the Asterisk
server at location 2 to
send all it outbound calls to the Asterisk Server at location 1.
Both asterisk servers can dial each other using extensions without a
problem, but when
users on Asterisk server 2, dial 9XXX-XXX- the call never reaches
the zap channel
on Asterisk server 1.
I have a workaround working right now using switch = but I think there
should be a better
way to do this.
Asterisk Server 2 -- extensions.conf
[outbound]
exten = _9XXX,1, Dial(SIP/outbound-server/${EXTEN:1},30,r)
exten = _9XXX,2, HangUp()
Asterisk Server 2 -- sip.conf
[outbound-server]
type=friend
username=outbound-server
secret=
context = all-calls
host=1.1.1.1
nat=no
canreinvite=no
insecure=very
qualify=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
any help is accepted and appreciated.
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Re: [asterisk-users] HPEC audio clipping
- Noah Miller [EMAIL PROTECTED] wrote: 1. A patch allowing capture audio streams in a way that will allow [us] to debug (and presumably fix) the problem was mentioned by Kevin. - Anything new about it ? I couldn't find it in Zaptel 1.2.17.1 nor 1.4.2.1 changelog. I believe it was introduced in Zaptel 1.2.17.1. From the description of zaptel 1.2.17.1 posted to www.asterisk.org: Added the ability to monitor pre-echo cancellation audio with ztmonitor - Noah Yes, you are correct. -- Jason Parker Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Responding to SIP OPTIONS
Joshua Colp wrote: Alex Lake wrote: I understand that it is customary for SIP User Agents to send OPTIONS packets every now and then to check that a peer is still alive and well. Indeed I understand that Asterisk itself sends them if qualify is set to yes in the peer configuration. How is one supposed to configure the dialplan so that Asterisk responds correctly to these requests? At the moment, I'm seeing Looking for s in default and then a 404 Not Found being returned - which can't be right. Thanks! Alex Lake DIGITAL MAIL LIMITED Handling of OPTIONS in Asterisk has changed a little bit through chan_sip versions... but for the most part the other side usually just wants you to respond with something/anything. Is the other side unhappy with the 404 Not Found? Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users In general, a 4XX reply to a successful request is NEVER considered good form. N. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Phone and Server Behind NAT
Chris Shipman wrote: I have an asterisk Server (2.1.17) behind NAT with a static IP and port forwarding enabled. The remote SIP phone is also behind NAT. I've gotten them to work except when I specify a secret.When there is a secret configured the phone can not authorize. Has anyone gotten this to work? Regards, Chris I've seen this happen with some SIP aware solutions. They try to fix up the headers with the public IP address and end up screwing up the authentication hash so that it doesn't match when computed on the other side. What is doing the NAT? Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 - Part cut
Hi all, We are an ISP in Switzerland and we propose VoIP with Asterisk. Everything works perfectly for all clients but one. In a conversation, they have no sound during 2 to 8 seconds using the G729 codec (I didn't make the test with G711). The Client configuration is perfect (QoS and bandwidth management). Do you know some issues with the G729 codec? Thanks a lot for your comments, Thomas Ps: The client doesn't use all his bandwidth... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GXV-3000 IP Video Phone
Hello, So far yes... The Video phones are behaving good and all the functionality working. I have 5 phone on the network and planning to put more by next week. Cheers, Nitesh Noah Miller wrote: Hi Nitesh - Thanks everyone... The GXV-3000 IP Video Phone works with Asterisk 1.2 using H.263 Video Coder. I had to update both phones firmware with new one... Out of curiosity - do you like the phone? I've looked for reviews, but I haven't found any that rate the phone's functionality. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug in voicemail module of Asterisk 1.4.2?
Joshua Colp wrote: The voicemail email gets handed off to sendmail for actual sending. It's adding on the envelope above. Yes, but asterisk is writing the From: header. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Responding to SIP OPTIONS
SIP wrote: Joshua Colp wrote: Handling of OPTIONS in Asterisk has changed a little bit through chan_sip versions... but for the most part the other side usually just wants you to respond with something/anything. Is the other side unhappy with the 404 Not Found? Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users In general, a 4XX reply to a successful request is NEVER considered good form. Especially as there are other suitable empty reply codes. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] G729 - Part cut
Turn off VAD (Voice Activation Detection) on the client software or your carrier is using VAD. Asterisk does not like VAD Best regards Erick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Deillon Sent: Tuesday, May 08, 2007 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] G729 - Part cut Hi all, We are an ISP in Switzerland and we propose VoIP with Asterisk. Everything works perfectly for all clients but one. In a conversation, they have no sound during 2 to 8 seconds using the G729 codec (I didn't make the test with G711). The Client configuration is perfect (QoS and bandwidth management). Do you know some issues with the G729 codec? Thanks a lot for your comments, Thomas Ps: The client doesn't use all his bandwidth... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.2 tanking CPU
Using a quad core 1.86GHz Xeon CPU here, running Asterisk 1.4.2. Noticed the following: Cpu(s): 4.3% us, 95.4% sy, 0.0% ni, 0.2% id, 0.0% wa, 0.0% hi, 0.0% si 30908 asterisk 18 0 188m 10m 5152 S 400 0.3 51051:13 asterisk Asterisk is eating up all the cores running the CPU at 400%. Is there something broken in 1.4.2 that needs to be addressed? Any suggestions? There's currently zero calls going on. - sf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: h323 problem with asterisk 1.2.18
Hi guys, I had the same problem ... and then remembered that my asterisk 1.2.9.1 compiled just fine ... So, i tried that Makefile ... and voila! :) See attached patch ... Cesc On 5/8/07, nik600 [EMAIL PROTECTED] wrote: On 5/7/07, nik600 [EMAIL PROTECTED] wrote: i am experiencing problem with asterisk 1.2.18 I've downloaded and installed pwlib and openh323 with the following commands: cd /path/to/pwlib ./configure make clean opt cd /path/to/openh323 ./configure make clean opt then 'ive set the corresponding PATH PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/ export PWLIBDIR OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/ export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH but when i go to: cd asterisk-1.2.18/channels/h323/ and do a make opt: [EMAIL PROTECTED]:/data/programmi/asterisk_1.2.18/asterisk-1.2.18/channels/h323# make opt make: *** No rule to make target `opt'. Stop. why? where am i wrong? i've also tried the last version of pwlib and openh323, but without fixing the problem thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser i've also tried supported version Open H.323 version v1.17.1, PWLib v1.9.0 but.. it doesn't compile. It seems to be a problem with makefile -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users asterisk.1.2.18.svn63330.h323.patch Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
ax. The downside of rx_fax is that you need to compile it into asterisk. The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. The last fax setup I did was for a small 2-person office where they had an existing fax machine that answered, listened for the remote fax squawk, if it didn't get it, then it rung the phones daisy-chained to it, and if they didn't answer it went to answering machine. I implemented this in asterisk fairly easilly with rx_fax. I'm not sure if you can do that with iaxmodem. Another question along these lines : How does everyone one fax detection on a sip channel? The only thing I've found is NvFaxDetect - anyone know of anything else? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
The downside of rx_fax is that you need to compile it into asterisk. The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. The last fax setup I did was for a small 2-person office where they had an existing fax machine that answered, listened for the remote fax squawk, if it didn't get it, then it rung the phones daisy-chained to it, and if they didn't answer it went to answering machine. I implemented this in asterisk fairly easilly with rx_fax. I'm not sure if you can do that with iaxmodem. Another question along these lines : How does everyone one fax detection on a sip channel? The only thing I've found is NvFaxDetect - anyone know of anything else? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2 from svn ... lock on shutdown
Hi, I will add the report ... though I find the system a bit cumbersome for sporadic users like me. Oh, and you are right ... without chan_h323 asterisk shuts down just fine. Regards, Cesc On 5/8/07, Joshua Colp [EMAIL PROTECTED] wrote: Cesc wrote: Hi, I hope this gets picked up by some bug marshall ... Eep! Filing a bug is best instead of email it here for future reference... I attached gdb to the locked process: 0xb725af28 in std::_Rb_tree_increment () from /usr/lib/libstdc++.so.6 (gdb) bt #0 0xb725af28 in std::_Rb_tree_increment () from /usr/lib/libstdc++.so.6 #1 0xb793f304 in std::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::operator++ () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #2 0xb79881a0 in std::__distancestd::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #3 0xb79881cb in std::distancestd::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #4 0xb7989ee6 in std::_Rb_treePString, std::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase*, std::_Select1ststd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* , std::lessPString, std::allocatorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::erase () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #5 0xb7989f20 in std::mapPString, PFactoryOpalMediaFormat, PString::WorkerBase*, std::lessPString, std::allocatorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::erase () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #6 0xb7989f5a in PFactoryOpalMediaFormat, PString::Unregister_Internal () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #7 0xb7989f9d in PFactoryOpalMediaFormat, PString::Unregister () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #8 0xb7989fc9 in OpalPluginMediaFormat::~OpalPluginMediaFormat () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #9 0xb748bea1 in PAbstractList::RemoveAt () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #10 0xb74892e1 in PCollection::RemoveAll () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #11 0xb7489e25 in PAbstractList::DestroyContents () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #12 0xb7490152 in PContainer::Destruct () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #13 0xb791ca57 in PAbstractList::~PAbstractList () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #14 0xb79755c9 in PListOpalMediaFormat::~PList () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #15 0xb79828e7 in H323PluginCodecManager::~H323PluginCodecManager () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #16 0xb7e0d4f0 in exit () from /lib/tls/libc.so.6 #17 0x080bddd7 in quit_handler (num=135324439, nice=3, safeshutdown=1, restart=0) at asterisk.c:945 #18 0x080be019 in handle_shutdown_now (fd=1, argc=2, argv=0xb830) at asterisk.c:1104 #19 0x0809811b in ast_cli_command (fd=1, s=0x8151900 \001) at cli.c:1364 #20 0x080c0d93 in main (argc=2, argv=0xbd84) at asterisk.c:1019 (gdb) This is definitely an issue with chan_h323 and OpenH323. If you don't load chan_h323 can you then shut down fine? If so please file a bug on bugs.digium.com and the individual who looks after that stuff will look at it. Thanks! Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma cards for sale
We have several Sangoma cards that we used during a transition time in our the replacement of our legacy voice system that we no longer need. Each of them saw about a month of service and are in good working order. We'd be happy to get 70% of retail for them. They are as follows: qty 2 A104D PCI w/ on board echo cancelation - $1500 each qty 1 A102D PCI w/ on board echo cancelation - $1050 each Please contact me off list if you are interested. - Jeremy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] app_txfax, app_rxfax
I modified chan_sip.c to turn on a dsp to do fax detect based on inband dtmf being selected. And when reading rtp if 'f' character shows up vector to fax extension Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Tuesday, May 08, 2007 12:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_txfax, app_rxfax ax. The downside of rx_fax is that you need to compile it into asterisk. The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. The last fax setup I did was for a small 2-person office where they had an existing fax machine that answered, listened for the remote fax squawk, if it didn't get it, then it rung the phones daisy-chained to it, and if they didn't answer it went to answering machine. I implemented this in asterisk fairly easilly with rx_fax. I'm not sure if you can do that with iaxmodem. Another question along these lines : How does everyone one fax detection on a sip channel? The only thing I've found is NvFaxDetect - anyone know of anything else? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Could two Asterisk servers connect through VPN
NM == Noah Miller [EMAIL PROTECTED] writes: NM If it helps at all, I read a study that said that SSL VPN's can NM actually help with jitter problems. So it might be preferable to NM implement something with OpenVPN (uses SSL) rather than an NM IPSec-based VPN. I found the link: Only if you use gold-plated connectors and oxygen-free copper. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: h323 problem with asterisk 1.2.18
many thanks for your help! i have used a makefile of a release 1.2.13 and now i've correctly compiled it. On 5/8/07, Cesc [EMAIL PROTECTED] wrote: Hi guys, I had the same problem ... and then remembered that my asterisk 1.2.9.1 compiled just fine ... So, i tried that Makefile ... and voila! :) See attached patch ... Cesc On 5/8/07, nik600 [EMAIL PROTECTED] wrote: On 5/7/07, nik600 [EMAIL PROTECTED] wrote: i am experiencing problem with asterisk 1.2.18 I've downloaded and installed pwlib and openh323 with the following commands: cd /path/to/pwlib ./configure make clean opt cd /path/to/openh323 ./configure make clean opt then 'ive set the corresponding PATH PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/ export PWLIBDIR OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/ export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH but when i go to: cd asterisk-1.2.18/channels/h323/ and do a make opt: [EMAIL PROTECTED]:/data/programmi/asterisk_1.2.18/asterisk-1.2.18/channels/h323# make opt make: *** No rule to make target `opt'. Stop. why? where am i wrong? i've also tried the last version of pwlib and openh323, but without fixing the problem thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser i've also tried supported version Open H.323 version v1.17.1, PWLib v1.9.0 but.. it doesn't compile. It seems to be a problem with makefile -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sound files
Hello, Can i identify the sound files that are played in the asterisk console? I defined the verbose to 100 but i can not see the sound files that are played in some situations... :( For example, I need to know what files are played for the message: Extension xxx is unavailable The goal is to translate that to Portuguese (pt_pt)... Thanks in advance, PS. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] random sections lost from call recording
We are using Record to monitor calls. We use this because it has the
option of a max time in it's call.
the problem is, and I'm not at all sure it is happening in record, the
recordings have sections of the conversation missing, sometimes. there
is not significant pattern as to the types of calls that are having this
problem. It appears quite random.
we are running this on asterisk 1.2.12.
the record command in the extensions.conf file looks like :
exten = 983,n,Record(/recordings/${CALLFILENAME}:wav|0| ${MAXDUR}|noanswer)
Thanks
Don Fletcher
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[asterisk-users] MYSQL Query -- PAGE
I have all my SIP users in a realtime database. I would like to use
MySQL command to query the database and use the results from the query
to page all the phones found in the query.
The results from the MySQL query will be multiple rows of extension:
Something like:
mysql Select extension from sip where extension like '6%'
6001
6002
6003
ex
I need to put all the results into a variable that would equal something like:
SIP/6001SIP/6002SIP/6003
I have setup a couple basic MYSQL Query's for my dialplan. Mostly
just looking up a DID to Extension Mapping for setting callerid on
outbound and inbound calls.
How does asterisk handle the multiple results. Is there a way to loop
until there are no more rows?
Something like Set(devices=${devices}${newrow_result})
I looked at the example on
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL but that doesn't
seem to be accurate.
Thanks all!!
--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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[asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 42
I will be out of the office until Monday, May 14. Please contact OWD at 800-337-3839 and ask for Client Services if you need immediate assistance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nagios/Cacti Plugin
Is this for asterisk 1.2 or asterisk 1.4? 2007/4/26, bkruse [EMAIL PROTECTED]: Hey guys, In my spare time(off of work, not digium related whatsoever) I finished the cacti php script. I need someone to help me do some finishing touches and make a basic layout and pretty colors for the template. All the grunt work and data sources are there, just need to put them into graphs and make them look nice and what not. If your interested in helping/doing this for me, email me at: [EMAIL PROTECTED] Thanks Guys! so far this plugin is Rockin! -bkruse ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sound files
On 5/8/07, Pedro Silva [EMAIL PROTECTED] wrote: Can i identify the sound files that are played in the asterisk console? I defined the verbose to 100 but i can not see the sound files that are played in some situations... :( For example, I need to know what files are played for the message: Extension xxx is unavailable The goal is to translate that to Portuguese (pt_pt)... Try this page: http://www.nathanpralle.com/software/ast_masterlist.html Not 100% up to date, but it covers most of the prompts I'd had to look up. -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vista compatibilty in SIP softphones
Greetings list, I've noticed over the last couple of weeks that, unsurprisingly, nearly every new PC seems to be coming with Vista these days. I expect it'll only be a matter of time for all of us before clients start needing Vista-compatible softphones (if it's not already happened). So, what's the story with Vista compatibility amongst the softphones currently out there? Ideally, I'd like to find a decent open-source Vista-compatible softphone, but free, even if closed-source would do the job for the time being. What are your experiences with SIP softphones under Vista? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL Query -- PAGE
Forrest Beck wrote:
I have all my SIP users in a realtime database. I would like to use
MySQL command to query the database and use the results from the query
to page all the phones found in the query.
The results from the MySQL query will be multiple rows of extension:
Something like:
mysql Select extension from sip where extension like '6%'
6001
6002
6003
ex
I need to put all the results into a variable that would equal something
like:
SIP/6001SIP/6002SIP/6003
I have setup a couple basic MYSQL Query's for my dialplan. Mostly
just looking up a DID to Extension Mapping for setting callerid on
outbound and inbound calls.
How does asterisk handle the multiple results. Is there a way to loop
until there are no more rows?
Something like Set(devices=${devices}${newrow_result})
I looked at the example on
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL but that doesn't
seem to be accurate.
Thanks all!!
What I've done in postgresql is to build an pl/pgsql procedure that
returns the desired dialstring. So the procedure does the select and
then concats them.
--
Remco Post
I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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Re: [asterisk-users] Vista compatibilty in SIP softphones
- Chris Bagnall [EMAIL PROTECTED] wrote: Greetings list, I've noticed over the last couple of weeks that, unsurprisingly, nearly every new PC seems to be coming with Vista these days. I expect it'll only be a matter of time for all of us before clients start needing Vista-compatible softphones (if it's not already happened). So, what's the story with Vista compatibility amongst the softphones currently out there? Ideally, I'd like to find a decent open-source Vista-compatible softphone, but free, even if closed-source would do the job for the time being. What are your experiences with SIP softphones under Vista? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons Zoa over at asteriskguru was kind enough to send me a beta version (which is now released) of idefisk v2, after I told him I was using Vista (yeah, yeah, I insaned for a day there when I got the CD from HP). It actually worked really well. There were some mic issues, but those were driver related. idefisk isn't open source, but there is a free version with a bunch of features. -- Jason Parker Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sangoma A101 on Freebsd 6.2
Hi Can you send me output from 'pciconf -l'? Thanks Alex Feldman Software Project Leader 905.474.1990 x104 _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zvonimir Mileta Sent: Tuesday, May 08, 2007 8:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sangoma A101 on Freebsd 6.2 I have an issue that hopefully you can help me solve. I've got the sangoma a101 card and installed it on freebsd but I according to the manual I should be see when running dmesg PCi0 vendor. something that tells me sangoma it's being recognized by the system. Now this is the 2nd card I try, the first one according to support was faulty and they sent me a new one. Is it possible to be doing something wrong? Is freebsd maybe recognizing differently from what it says on the manual? Please any help would be highly appreciated. -Zvonimir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem when PABX call to Asterisk by Unicall
Hi all, I have an Asterisk server connected in a PABX (TELEDATA) by channel Unicall.. I`m having problem when somebody call from PABX to Asterisk.. Eg: When somebody dial 1234, I received 113344 in the Asterisk CLI... If somebody can help me... or already saw this... Everton Goularth Uberlandia - MG - Brazil ___ Yahoo! Mail - Sempre a melhor opção para você! Experimente já e veja as novidades. http://br.yahoo.com/mailbeta/tudonovo/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL Query -- PAGE
Well This seems to work.
[macro-pageall]
; Context for paging all devices.
; This will search the sip table in the realtime database
; for all phones that start with a number. That number is
; passed to this macro as ${ARG1}.
;
; ARG1 = The first digit of the phones to be paged (US Campus=6,
LS Campus=2)
; ARG2 = Device for the PA system. If the user selected to
; page the PA system. That will be included.
;
exten = s,1,MYSQL(Connect connid ${realdb_host} ${realdb_user}
${realdb_pass} ${realdb_db})
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ name\ FROM\ sip\
WHERE\ name\ LIKE\ '${ARG1}%')
exten = s,3,MYSQL(Fetch fetchid ${resultid} number)
exten = s,4,GoToIf($[${fetchid} = 1]?5:8)
exten = s,5,Set(pagedevice=${pagedevice}SIP/${number})
exten = s,6,NoOp(${number})
exten = s,7,GoToIf($[${fetchid} = 1]?3:8)
exten = s,8,Set(pagedevice=${pagedevice:1})
exten = s,9,NoOp(PageDevice ${pagedevice})
exten = s,10,MYSQL(Clear ${resultid})
exten = s,11,MYSQL(Disconnect ${connid})
exten = s,12,GoToIf($[${ARG2} != ]?13:14)
exten = s,13,Set(pagedevice=${pagedevice}${ARG2})
exten = s,14,Set(_ALERT_INFO=RA)
exten = s,15,Page(${pagedevice})
exten = s,16,Hangup()
On 5/8/07, Remco Post [EMAIL PROTECTED] wrote:
Forrest Beck wrote:
I have all my SIP users in a realtime database. I would like to use
MySQL command to query the database and use the results from the query
to page all the phones found in the query.
The results from the MySQL query will be multiple rows of extension:
Something like:
mysql Select extension from sip where extension like '6%'
6001
6002
6003
ex
I need to put all the results into a variable that would equal something
like:
SIP/6001SIP/6002SIP/6003
I have setup a couple basic MYSQL Query's for my dialplan. Mostly
just looking up a DID to Extension Mapping for setting callerid on
outbound and inbound calls.
How does asterisk handle the multiple results. Is there a way to loop
until there are no more rows?
Something like Set(devices=${devices}${newrow_result})
I looked at the example on
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL but that doesn't
seem to be accurate.
Thanks all!!
What I've done in postgresql is to build an pl/pgsql procedure that
returns the desired dialstring. So the procedure does the select and
then concats them.
--
Remco Post
I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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[asterisk-users] voip-info.org mirrors?
Hi: It's been a few weeks since the great voip-info.org crash. Around that time there was some lofty talk about a set of mirrors being set up for it. Has anything happened with that, or are we just going back to business as usual? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ericsson dialog 4187
Hi, Anybody using this Ericcson analog phone with Asterisk: Ericsson dialog 4187? I was told some functionalities like CLID will only work with an Ericsson PABX but other than that I would like to hear from anybody using this phone on a FXS port. Thanks, Jose Limeres ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems witch SPA3102.
Hello, i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with cdr. Well all I want is to receive incoming calls from pstn on specified sip account (suppose 8000), and to initiate outgoing calls from all my asterisk sip accounts through SPA3102 device. Someone can explain me what may i set on SPA and asterisk to do this thing. Thank you for your support. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Could two Asterisk servers connect through VPN
How about required MTU and jitter? I think openvpn will add some latency and frames will be charged with supplementary encapsulation bits. On 08 May 2007 19:03:09 +0200, Benny Amorsen [EMAIL PROTECTED] wrote: NM == Noah Miller [EMAIL PROTECTED] writes: NM If it helps at all, I read a study that said that SSL VPN's can NM actually help with jitter problems. So it might be preferable to NM implement something with OpenVPN (uses SSL) rather than an NM IPSec-based VPN. I found the link: Only if you use gold-plated connectors and oxygen-free copper. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with the loading of the cards in Debian
Cohen, Thanks for your help, but I solved this problem removing the ACPI and APIC from the boot in /boot/grub/menu.lst. Thanks, MCelo. 2007/5/8, Tzafrir Cohen [EMAIL PROTECTED]: On Mon, May 07, 2007 at 05:15:26PM -0300, MCelo wrote: Cohen, On different boots you get the modules loaded with a different order? Yes, thats it. What do you have in /etc/modules ? This should take effect on boot. I have the following in /etc/modules : asterisk:~# cat /etc/modules # /etc/modules: kernel modules to load at boot time. # # This file contains the names of kernel modules that should be loaded # at boot time, one per line. Lines beginning with # are ignored. zaptel You don't really need zaptel here. It will get loaded by a modprobe of any of the other. wcte11xp wctdm wcfxo OK. loop This is the order that I want. I don't know what loop means. What do you have in /etc/sysconfig/zaptel ? This should take efect if you unloaded all modules and want to reload them. I don't have the file /etc/sysconfig/zaptel, but I have /etc/default/zaptel. Right. My mistake. Where I can find some information about the loading modules. In this file I un-commented the modules that I want, and I left commented the modules that I don't want. I have the following in /etc/default/zaptel : asterisk:~# cat /etc/default/zaptel TELEPHONY=yes #DEBUG=yes Removing remmed-out lines: MODULES=$MODULES wcte11xp # TE110P - Single Span T1/E1 Card MODULES=$MODULES wcfxo# X100P - Single port FXO interface MODULES=$MODULES wctdm# TDM400P - Modular FXS/FXO interface So you basically have: MODULES=$MODULES wct1xxp wcfxo wctdm Also note that you don't initialize MODULES . I believe /etc/init.d/zaptel has a default value for it. grep MODULES= /etc/init.d/zaptel -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ringing Volume
Hi, Does anyone know how to adjust the volume of the ringing application? I have done a lot of internet searching and have not found much. Thanks. Jad Network Blitz Bkgrd.gif___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing Volume
Jadrien Wauthier wrote: Does anyone know how to adjust the volume of the ringing application? I have done a lot of internet searching and have not found much. You cannot do this in Asterisk. Some SIP phones might allow you to do so by setting an option on the phone, but you would have to ask the company that makes that specific phone how to do that. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] LDAPget or something else?
Hi All, We are currently using LDAPget 1.0rc6 with Asterisk 1.2.x. I see that there is LDAPget 2.0rc1 for Asterisk 1.4.x. I was wondering if there was something better. Are people using LDAPget or something else? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.2 and UDP packet numbering on bridged channels (for jitter buffering)?
http://www.asterisk.org/node/48317 does a nice job of explaining the 1.4 jitter buffer, however it raised a question in my mind. In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the UDP RTP packets renumbered on transmit, or is the original sequence number preserved in the UDP header? A comment is made on the referenced blog that jitter buffering is best implemented at the endpoint, where additional jitter will not be added by another IP link. This is logical thinking, but only possible if the bridging function in Asterisk preserves the source call leg UDP packet numbering in the terminating call LEG UDP RTP packet stream. If the effect of the Asterisk SIP to SIP bridge is such that the UDP headers are re-created on transmit it is likely that the packet sequencing is the order in which Asterisk transmitted the packets, which is may not be the order in which the original source UA transmitted them due to jitter in the IP link on the first half of the bridged call. Can anyone provide an authoritative answer on how asterisk sequences UDP RTP packets on the transmit leg of a bridged SIP call (known based on actual testing or code review)? Or maybe there is information I lack that makes this a silly question, such as where the SIP RTP sequence number is stored in the packet (ie: not in the UDP header?) :-) Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL Query -- PAGE
Forrest Beck wrote:
How does asterisk handle the multiple results. Is there a way to loop
until there are no more rows?
I don't use Realtime in Asterisk personally so I'm not sure if it
implements it or not, but I agree that being able to iterate over a
ResultSet is a pretty basic need. I think I remember AEL2 being able to
do that. rushowr put together a nice collection of AEL2 scripts (link
below) that probably has something in it you could use. I know he uses
MySQL a lot in is dialplans. You could also use an AGI/FastAGI to do
something like that.
If you don't mind a small FastAGI listener running and you don't mind
Pascal, you could check out AsterPas (link below) which does support
doing that with MySQL, FirebirdSQL and Sqlite databases and its free
(though not open source). It's still considered beta, but we're using
it ourselves quite a bit without problems.
Also, there is Astersk Java (link below) which looks dynamite if you're
more familiar with or prefer Java.
Personally, I like the idea of pushing non-asterisk operations out of
Asterisk so AGI/FastAGI is my preference. Many also seem to advocate
using AEL2 which is pretty powerful and easy.
Asterisk Java:
http://asterisk-java.org
AsterPas:
http://www.datatrakpos.com/pos/datatalk/asterpas.aspx
SKeMAEL AEL2 Scripts from
With AsterPas, you could do something like the following:
{uses sqldb}
program BuildMyCrazyDialString;
Var
rowset: TDTRowset;
sDial: string;
begin
with SQLDB do
begin
SetProp('sqltype', 'sql');
SetProp('Connection', 'MyRealtimeDBConn');
SetProp('sql', 'SELECT xtenNumber FROM my_extensions_table WHERE ' +
'my_field = ' + AGI.GetVariable('MyGroupID'));
if (CreateRowSet('xtens')) then
begin
rowset := GetRowSet('xtens');
while (not rowset.eof) do
begin
if (sDial = '') then
sDial = 'SIP/' + rowset.AsString('xtenNumber')
else
sDial := sDial '' + 'SIP/' + rowset.AsString('xtenNumber');
rowset.Next;
end;
RemoveRowset('xtens');
end;
end;
// push out the result to the CLI
AGI.Noop('The DialString is: ' + sDial);
// set a dialplan variable for use when the FastAGI exits
AGI.SetVariable('DialStrReturn', sDial);
end.
--
Warm Regards,
Lee
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Re: [asterisk-users] MYSQL Query -- PAGE
Lee Jenkins wrote: Forrest Beck wrote: How does asterisk handle the multiple results. Is there a way to loop until there are no more rows? Sorry, I forgot the last link for the AEL2 scripts: http://sourceforge.net/projects/aelscriptlib/ -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP peer / Maximum retries exceeded on transmission
(repost - can anyone confirm whether they've seen this before, or have any tipes in debugging it?) Hi Everyone, I was hoping someone might know why I am experiencing a problem with Asterisk logging the event: [May 3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 669371069 (Critical Response) This is happening after: - call is setup, 2 way audio - call can function correctly for up to 5 minutes, with the external provider re-inviting every 1 minute When the problem happens - external peer re-invites asterisk - asterisk sends 200 OK - external peer sends ACK - asterisk retransmits 200 OK - external peer sends ack - .. - asterisk retransmits 200 OK (Retransmitting #6) - external peer sends ack - Asterisk logs the above message about maximum retries exceeded, and sends BYE to the inside SIP UA. The network configuration is as follows: phone -- alternative SIP server -- Asterisk -NAT- External peer The alternative SIP server is not a B2BUA, just SIP proxy. Now, sometimes a call can work without any problems, but not as often as when the above symptoms are experienced. The references I've found online about this type of problem suggest NAT as being the culprit, but in this case, Asterisk is logging it's reception of the ACK but deciding to ignore it and retransmit the 200 OK anyhow. I'm guessing in other cases people suspect is' NAT because they believe SIP isn't getting back trhough after a period of time. I was using 1.4.2, but found this changelog today for 1.4.3: ftp://ftp.digium.com/pub/asterisk/releases/ChangeLog-1.4.3 2006-09-30 16:12 + [r44068-44078] Paul Cadach [EMAIL PROTECTED] * channels/chan_sip.c: Found some buggy SIP clients (phones Planet VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK not on OK message only (when remote party answers) but on RINGING message too, so when we send 200 OK message, we get unidentified ACK message (because INVITE acknowledged on RINGING message already), so 200 OK retransmits within its retransmission interval then call gets dropped. If someone else knows how to provide workaround for such cases, please, fix it in correct way. Thanks to ssh from #asteriskru for provide access to his box to study and fix this case. I've upgraded to 1.4.4 but the problem still persists. The above changelog doesn't sound exactly like what Im experiencing but maybe it's related. Attached is my sip.conf, extensions.conf, and (debug = 10) logs for one example. I don't know what else might be needed to help anyone assist me in this problem - let me know if I missed something. It *feels* like an Asterisk bug but maybe a SIP expert can spot the problem in signalling/RFC conformance.. Thanks in advance, Chris Bennett [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=proxy.myhostname disallow=all allow=alaw sipdebug = yes recordhistory=yes dumphistory=yes register = authstuff@sip.externalpeer.com externhost=proxy.myhostname localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 localnet=169.254.0.0/255.255.0.0 nat=never canreinvite=no [authentication] auth = authstuff@sip.externalpeer.com [provider] type=peer username=myusername secret=mysecret fromuser=myusername fromdomain=sip.externalpeer.com host=sip.externalpeer.com nat=never canreinvite=no [] type=friend username= secret=secret host=dynamic context=tutorial nat=never insecure=invite qualify=yes [general] static=yes writeprotect=no clearglobalvars=no [globals] CONSOLE=Console/dsp [tutorial] exten = _XXX.,1,Dial(SIP/[EMAIL PROTECTED],,r) asterisk.logs.example1.txt.bz2 Description: BZip2 compressed data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2 and UDP packet numbering on bridged channels (for jitter buffering)?
Damon Estep wrote: http://www.asterisk.org/node/48317 does a nice job of explaining the 1.4 jitter buffer, however it raised a question in my mind. In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the UDP RTP packets renumbered on transmit, or is the original sequence number preserved in the UDP header? A comment is made on the referenced blog that jitter buffering is best implemented at the endpoint, where additional jitter will not be added by another IP link. This is logical thinking, but only possible if the bridging function in Asterisk preserves the source call leg UDP packet numbering in the terminating call LEG UDP RTP packet stream. If the effect of the Asterisk SIP to SIP bridge is such that the UDP headers are re-created on transmit it is likely that the packet sequencing is the order in which Asterisk transmitted the packets, which is may not be the order in which the original source UA transmitted them due to jitter in the IP link on the first half of the bridged call. Can anyone provide an authoritative answer on how asterisk sequences UDP RTP packets on the transmit leg of a bridged SIP call (known based on actual testing or code review)? I can tell you about our extensive tests back when we were on version 1.0.X Asterisk would take in an RTP stream and then recreate a new one on exit, putting in a new Sequence Number, and new Timestamp in the RTP Header. This effectly destroys any chance of efficiently relying on jitter buffering at the endpoints. From multiple tests over the years we have come to rely on the best jitter buffer we could devise in Asterisk regarding SIP-SIP channels. That is we loop the call out to a ZAP channel and back in, thus turning the call into SIP-ZAP-ZAP-SIP. The ZAP channels have quite good jitter buffers and they work perfectly in our configuration. Sure you eat extra T1 channels but we have not choice. Most of our customers are overseas and the jitter is quite high. Or maybe there is information I lack that makes this a silly question, such as where the SIP RTP sequence number is stored in the packet (ie: not in the UDP header?) J Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Aastra phones?
Sorry for being a little off topic, but I'mconsidering a few new phones for my Asterisk installation. I have a mix of Polycom 500/600s and an Aastra 480i CT. I'm considering adding a couple of Aastra 57i or 57i CT. Does anyone here have experience with the 480i CT and the newer 57i CT? I'm curious as to the real differences. Thanks, Michael Graves Sr Product Specialist Pixel Power Inc [EMAIL PROTECTED] [EMAIL PROTECTED] FWD 54245 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ringing Volume
Does anyone know how to adjust the volume of the ringing application? I have done a lot of internet searching and have not found much. You cannot do this in Asterisk. Some SIP phones might allow you to do so by setting an option on the phone, but you would have to ask the company that makes that specific phone how to do that. If Asterisk generates the audio, then it seems that there would be a source file that I could edit if nothing else. I looked at the app_dial.c, but I didn't see anything. Maybe I over looked something. If I lower the volume on the phone, then all audio on the phone would be lower. I am just interested in lowering the volume of the ringing. Basically, rings from the pstn is at one level, and the rings from Asterisk are at another level. I need to normalize the Asterisk volume. Thank you so much for your help with this. Jad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vista compatibilty in SIP softphones
I have Vista on my new HP laptop X-lite soft phone works like charm with it,
I tried sjphone, I couldnt get that working, its gets hung.
--
Deepak
Chris Bagnall [EMAIL PROTECTED] wrote:
Greetings list,
I've noticed over the last couple of weeks that, unsurprisingly, nearly every
new PC seems to be coming with Vista these days. I expect it'll only be a
matter of time for all of us before clients start needing Vista-compatible
softphones (if it's not already happened).
So, what's the story with Vista compatibility amongst the softphones currently
out there? Ideally, I'd like to find a decent open-source Vista-compatible
softphone, but free, even if closed-source would do the job for the time being.
What are your experiences with SIP softphones under Vista?
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons
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[asterisk-users] Re: Call waiting tone
Hello,
A few days ago I've asked about the ability to play a stuttered ringing
tone when the called party is already on the phone. I've found a partial
solution for it.
To describe again the problem: When a user is on a call and someone else
calls him, the caller does not know that the called party is on the phone
(while the called party wants to know that someone else is calling him/her and
not just play busy). On our public PSTN the caller is notified by a stuttered
ringing tone (thus he can decide whether to wait or hangup and call later).
What's I've done is that when the called party is on the phone a short message
is sent to the caller (may be also a recording of the stuttered ringing) and
then the call is passed. Here is the code fragment:
exten = _806XX,n,Set(Status=${DEVSTATE(SIP/${EXTEN})}) ; Get his status
exten = _806XX,n,GotoIf($[${Status} == NOT_INUSE]?OK:WAITING_CALL)
exten = _806XX,n(WAITING_CALL),Playback(waiting-call)
exten = _806XX,n(OK),Dial(SIP/${EXTEN}${aEXTEN},20,) ; Dial the phone for 20
Hope it helps someone...
Regards, __Yehavi:
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Re: [asterisk-users] Responding to SIP OPTIONS
Hi Alex, How is one supposed to configure the dialplan so that Asterisk responds correctly to these requests? At the moment, I'm seeing Looking for s in default and then a 404 Not Found being returned - which can't be right. Not specific to an OPTIONS packet, but I know that I previously experienced wierdness when I had my dialplan matching too much. For me it was a 'default route' for all calls going out a particular SIP peer. exten = _.,1,Dial(SIP/[EMAIL PROTECTED],,r) There were instances of SIP reinvites that would match this dial plan and be dialled back out to the provider. My fix was _ etc, matching more specifically what extensions I wanted to dial out to that provider. Your problem looks similar - Asterisk, based on your dialplan is initerpreting the special extension s as some dial attempt, resulting in 404 Not Found. There are a bunch of these special extensions: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf Predefined Extension Names Asterisk uses some extension names for special purposes: * i : Invalid * s : Start * h : Hangup * t : Timeout * T : AbsoluteTimeout * o : Operator This is my guess anyhow - if this isn't right, hopefully someone else can pin it down for you .. Regards, Chris Bennett ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP peer / Maximum retries exceeded on transmission
I can confirm the same error message... i haven't done nearly the amount of debuggin you have but it's the exact same error message i receive when i use a software based SIP phone connecting to another internal software SIP phone... some times it's twinkle to xlite some times xlite to xlite and some times twinkle to twinkle ... That's about all that i can confirm :) hope you get some help cuz i was also looking for some info on what the issue was. On 5/8/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: (repost - can anyone confirm whether they've seen this before, or have any tipes in debugging it?) Hi Everyone, I was hoping someone might know why I am experiencing a problem with Asterisk logging the event: [May 3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 669371069 (Critical Response) This is happening after: - call is setup, 2 way audio - call can function correctly for up to 5 minutes, with the external provider re-inviting every 1 minute When the problem happens - external peer re-invites asterisk - asterisk sends 200 OK - external peer sends ACK - asterisk retransmits 200 OK - external peer sends ack - .. - asterisk retransmits 200 OK (Retransmitting #6) - external peer sends ack - Asterisk logs the above message about maximum retries exceeded, and sends BYE to the inside SIP UA. The network configuration is as follows: phone -- alternative SIP server -- Asterisk -NAT- External peer The alternative SIP server is not a B2BUA, just SIP proxy. Now, sometimes a call can work without any problems, but not as often as when the above symptoms are experienced. The references I've found online about this type of problem suggest NAT as being the culprit, but in this case, Asterisk is logging it's reception of the ACK but deciding to ignore it and retransmit the 200 OK anyhow. I'm guessing in other cases people suspect is' NAT because they believe SIP isn't getting back trhough after a period of time. I was using 1.4.2, but found this changelog today for 1.4.3: ftp://ftp.digium.com/pub/asterisk/releases/ChangeLog-1.4.3 2006-09-30 16:12 + [r44068-44078] Paul Cadach [EMAIL PROTECTED] * channels/chan_sip.c: Found some buggy SIP clients (phones Planet VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK not on OK message only (when remote party answers) but on RINGING message too, so when we send 200 OK message, we get unidentified ACK message (because INVITE acknowledged on RINGING message already), so 200 OK retransmits within its retransmission interval then call gets dropped. If someone else knows how to provide workaround for such cases, please, fix it in correct way. Thanks to ssh from #asteriskru for provide access to his box to study and fix this case. I've upgraded to 1.4.4 but the problem still persists. The above changelog doesn't sound exactly like what Im experiencing but maybe it's related. Attached is my sip.conf, extensions.conf, and (debug = 10) logs for one example. I don't know what else might be needed to help anyone assist me in this problem - let me know if I missed something. It *feels* like an Asterisk bug but maybe a SIP expert can spot the problem in signalling/RFC conformance.. Thanks in advance, Chris Bennett ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk with festival facing problem
hi List, i've been trying to get festival work on my 1.4.4 *box for the last 3days, i've used the tutorial on this page http://www.voip-info.org/wiki-Asterisk+Festival+installation with exactly the same line in my dialplan just to make a test now when i try to call( dial 555 ) from my softphone i get this message on festival server debugger: serverTue May 8 11:36:53 2007 : Festival server started on port 1314 client(1) Tue May 8 11:37:31 2007 : accepted from localhost.localdomain client(1) Tue May 8 11:37:31 2007 : disconnected then from my CLI there nothing after parsing '/etc/asterisk/festival.conf' : found and my softphone get connected and can stay so till i hang up without any sound did someone esperienced this situation??? any clue?? thanks in advance -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
