Re: [asterisk-users] G.729 codec in pass-thru mode
Matthew, allow=all is defined in sip.conf for the ITSP's SIP peer. Additionally, ITSP supports g729 codec as we are able to send the traffic from other soft switch. In case g729 on asterisk box, as I mentioned earlier, call even doesn't go out of the asterisk box. Below extracts from log also indicate the same thing. [Jun 5 12:46:49] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456) [Jun 5 12:46:49] == Using SIP RTP CoS mark 5 [Jun 5 12:46:49] -- Couldn't call yyy.yyy.yyy.yyy/12127773456 [Jun 5 12:46:49] == Everyone is busy/congested at this time (0:0/0/0) Regards, Kamlesh Date: Tue, 4 Jun 2013 10:27:11 -0500 From: mr...@imminc.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G.729 codec in pass-thru mode Kamlesh Kumar wrote: SIP.conf [100] username=100 secret=password type=friend host=dynamic nat=yes canreinvite=no insecure=port disallow=all allow=ulaw allow=alaw allow=g729 context=asterisk qualify=no Is there also an allow=g729 line in sip.conf for the ITSP's SIP peer? SIP Trace: 201.xxx.xxx.xxx = SIP Softphone which originates the call xxx.xxx.xxx.xxx = Asterisk server yyy.yyy.yyy.yyy = ITSP ... --- SIP read from UDP:yyy.yyy.yyy.yyy:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060 From: 100 sip:1...@xxx.xxx.xxx.xxx;tag=as643c20b1 To: sip:12127773...@yyy.yyy.yyy.yyy;tag=gK029aaa8c Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 102 INVITE Contact: sip:12127773...@yyy.yyy.yyy.yyy:5060 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Content-Length: 234 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy s=SIP Media Capabilities c=IN IP4 zzz.zzz.zzz.zzz t=0 0 m=audio 21996 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 - [Jun 3 13:11:31] --- (11 headers 11 lines) --- [Jun 3 13:11:31] Found RTP audio format 0 [Jun 3 13:11:31] Found RTP audio format 101 [Jun 3 13:11:31] Found audio description format PCMU for ID 0 [Jun 3 13:11:31] Found audio description format telephone-event for ID 101 [Jun 3 13:11:31] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jun 3 13:11:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 3 13:11:31] Peer audio RTP is at port zzz.zzz.zzz.zzz:21996 [Jun 3 13:11:31] -- SIP/yyy.yyy.yyy.yyy-34d9 is making progress passing it to SIP/100-34d8 [Jun 3 13:11:31] Audio is at xxx.xxx.xxx.xxx port 26042 [Jun 3 13:11:31] Adding codec 0x4 (ulaw) to SDP [Jun 3 13:11:31] Adding non-codec 0x1 (telephone-event) to SDP This response from the ITSP says that only u-law may be used for the call. Please contact the ITSP and confirm that they actually support the G.729 codec. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] incoming DAHDI Channel explained
Hi, I use a Sangoma A104d-Card (with 4 x germany E1). I process some calls via an AGI-Script. When parsing the AGI-Variables I can see one that look like that: [agi_channel] = DAHDI/i3/211123456-89c What hat do the values mean in detail, please? DAHDI : this is clear i3 : does it mean, that the call comes in via E1-Port 3? 211123456 : Incoming-Call Caller-ID -89c : ? WANPIPE Release: 7.0.1 DAHDI Version: 2.6.2 Echo Canceller: HWEC libpri version: 1.4.12 Best regards -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sendmail when no response
hello list, i need your help please regarding send mail i use astreisk 1.4; i try to send mail when no response like below exten = 5xx,1,Dial(SIP/223, 10) exten = 5xx,n,system(echo test ${DNIS} Email| mail -s 'Call failed' myadresseem...@gmail.com) when i launch the CLI i found : You have new mail in /var/spool/mail/root i check the root and i found : Return-Path: root Received: (from root@localhost) by localhost.localdomain (8.13.1/8.13.1/Submit) id r55B3Deh023821; Wed, 5 Jun 2013 11:03:13 GMT Date: Wed, 5 Jun 2013 11:03:13 GMT From: root root Message-Id: 201306051103.r55B3Deh023821@localhost.localdomain To: failed, myadresseem...@gmail.com Subject: Call test Email --r55B3Dei023821.1370430193/localhost.localdomain-- could you please tell me how to do in order to send email to my address gmail for example thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sendmail when no response
Your problem looks more like an MTA configuration problem. You need at least a valid relay host. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming DAHDI Channel explained
Sangoma's tech support is probably the better source of information. DAHDI: obviously DAHDI channel i: incoming call 3: span 3 (not the port) 211123456: CLID, probably subject to filtering (see national/international prefix settings) 89c: internal counter (i.e. 2204 calls so far) jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 codec in pass-thru mode
Kamlesh Kumar wrote: allow=all is defined in sip.conf for the ITSP's SIP peer. Additionally, ITSP supports g729 codec as we are able to send the traffic from other soft switch. There must be some difference between your Asterisk servers. Please set them up for calling the ITSP with G.729 and provide the following CLI output from both of them. Be sure to preserve any differences when obscuring IP addresses and label the output clearly as G.729 working and G.729 fails. CLI core show version CLI sip show settings CLI sip show peer 100 CLI sip show peer ITSP's SIP peer CLI g729 show licenses Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming DAHDI Channel explained
Sangoma's tech support is probably the better source of information. DAHDI: obviously DAHDI channel i: incoming call The 'i' is for ISDN not incoming call since it will be this way for outgoing calls as well. 3: span 3 (not the port) 211123456: CLID, probably subject to filtering (see national/international prefix settings) 89c: internal counter (i.e. 2204 calls so far) The other fields are pretty much as described by jg. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming DAHDI Channel explained
Yes, my assumption was wrong and to make things worse, my CDR data clearly show that i cannot denote incoming calls. Maybe it's time that I learn the rules as well: Analog channels do not seem to have a special identifier. The 1st call for analog channel 13 would be s.th. like DAHDI/13-1. Outside calls via an ISDN connection with s.th. like DIAL(DAHDI/r2/08932168,..) would dial the number using DAHDI group 2 in a round robin fashion, but internally the channel would be s.th. like DAHDI/iX/08932168-abcd. The span X is not related to the dial group and depends on the configuration. For a BRI device a single span has 2 channels, a PRI device up to 30. As far as channel variables go the actual channel does not seem to get reported, but this is not really necessary. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming DAHDI Channel explained
B.H. Hi! On Wed, Jun 5, 2013 at 7:26 PM, jg webaccou...@jgoettgens.de wrote: For a BRI device a single span has 2 channels, a PRI device up to 30. As far as channel variables go the actual channel does not seem to get reported, but this is not really necessary. AFAIK, at least for AMI listeners, the real channel/span is reported by DAHDIChannelEvent attributes: 'dahdichannel' reports the actual DAHDI channel number 'dahdispan' is a span number jg -- משיח NOW! Moshiach is coming very soon, prepare yourself! יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem to install asterisk on vps digitalocean
2013/6/4 James Cloos cl...@jhcloos.com mailto:cl...@jhcloos.com t == troxlinux xserverli...@gmail.com mailto:xserverli...@gmail.com writes: t I try to install asterisk on vps server , but fails when I want to t install dahdi There is no hardware for dahdi to use; you shouldn't need to install it. 2013-06-04 22:28, troxlinux skrev: if it is true I have not any hardware but I need help to solve it and I think it could serve other future Depending on the type of VPS this can work or not. My general experience of normal stock-VPS:es are that they are often oversold. And it will create a lot of issues for you with voip. On OpenVZ you can't load kernel-modules in the VE anyway, so it's no idea to compile Dahdi. But if you control the HN, you can install it there and give the VE access to it. In that case you should compile it in the VE as well. The same applies to LXC. On XEN/KVM I think it will work, but be sure to test the timing with dahdi_test - you don't want the results to drop below 99.9. Try create some i/o (for example compile asterisk) and run the test in another window. Dahdi in VE's is a pain, but can work if you control the HN. You should do development and testing on a dedicated server. After that, try move to a VPS and try to fix the issues. The other way around is very hard. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk HA
I was go through'ing the following links for HA, https://wiki.asterisk.org/wiki/display/TOP/Failover+-+Linux - which doesn't have file syncing. https://www.johncahill.net/wiki/index.php/2_Node_Active/Passive_cluster - this one has file syncing with pacemaker Any other HA applications available or the lsyncd with pacemaker is good? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users