[asterisk-users] Linux call router
hello there, I am facing an issue with misd/misdnuser/lcr in the system I am running debian 7 and I managed to install from git misdn/misdnuser but in lcr I am getting: chan_lcr.c: In function 'load_module': chan_lcr.c:3520:24: warning: assignment makes pointer from integer without a cast [enabled by default] make[2]: *** [chan_lcr.po] Error 1 make[2]: Leaving directory /usr/src/lcr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory /usr/src/lcr' make: *** [all] Error 2 root@voyage:/usr/src/lcr# could someone help me please? -- _ -- 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 Authentication
Hi, I am trying to setup asterisk so that anyone from any IP can call using any callerid as long they have an account - also no registration is required. However, it seems like asterisk tries to find peer based on either the IP address or from header. What I really want is asterisk to find account/peer based on username passed as part of the authentication and NOT from the IP address or the from header. Any idea how to achieve this. Thanks -- _ -- 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] Asterisk Authentication
On 11 March 2014 11:39, Jim Boykin boykin...@gmail.com wrote: Hi, I am trying to setup asterisk so that anyone from any IP can call using any callerid as long they have an account - also no registration is required. However, it seems like asterisk tries to find peer based on either the IP address or from header. What I really want is asterisk to find account/peer based on username passed as part of the authentication and NOT from the IP address or the from header. Any idea how to achieve this. Thanks It has to be either fixed IP address or username and password with a dynamic host. This is no in between to the best of my knowledge. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833
Hello, I have installed the latest version 12 that has been released (12.1.0.rc3). I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working. Is it a known issue? Below you can see the output of the asterisk monitor. --- Received SIP request (1182 bytes) from UDP:10.25.153.150:5060 --- INVITE sip:039988120@172.16.60.160:5060;user=phone SIP/2.0 Record-Route: sip:10.25.153.150;lr;ftag=02e3a8c0-33807b-t-2 Via: SIP/2.0/UDP 10.25.153.150:5060;branch=z9hG4bK587.67258295.0 Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuqC93X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJ3lqwWxarW.gqWReJMEPA36juW6WBzR363RVA3Ejugx3* Max-Forwards: 68 From: 39937841 39937841 sip:39937841;cpc=payphone@192.168.225.2:5060 ;user=phone;tag=02e3a8c0-33807b-t-2 To: sip:D39539988120@192.168.225.2:5060;user=phone Call-ID: 2915b6e4-02e3a8c0-be53@192.168.225.2 CSeq: 2 INVITE Contact: sip:10.1.1.10;line=sr-N6IAzBMsz.MwzxPfPxFsMJZfWBc7MBVuOBV-W.y6MxV* User-Agent: NetCentrex CCS Softswitch/7.16.0 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, INFO, PRACK, UPDATE, NOTIFY Supported: 100rel P-Asserted-Identity: 39937841 39937841 sip:39937841;cpc=payphone@192.168.225.2:5060;user=phone Min-SE: 90 Privacy: none Content-Type: application/sdp Content-Length: 167 v=0 o=10.206.22.171 62708 2 IN IP4 10.206.22.171 s=SIP Call c=IN IP4 10.206.22.171 t=0 0 a=sendrecv m=audio 41040 RTP/AVP 8 a=rtpmap:8 PCMA/8000/1 a=ptime:20 --- Transmitting SIP response (602 bytes) to UDP:10.25.153.150:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.25.153.150:5060 ;rport;received=10.25.153.150;branch=z9hG4bK587.67258295.0 Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuqC93X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJ3lqwWxarW.gqWReJMEPA36juW6WBzR363RVA3Ejugx3* Record-Route: sip:10.25.153.150;lr;ftag=02e3a8c0-33807b-t-2 Call-ID: 2915b6e4-02e3a8c0-be53@192.168.225.2 From: 39937841 39937841 sip:39937841;cpc=payphone@192.168.225.2 ;user=phone;tag=02e3a8c0-33807b-t-2 To: sip:D39539988120@192.168.225.2;user=phone CSeq: 2 INVITE Content-Length: 0 -- Executing [039988120@from-external:1] NoOp(PJSIP/sipp-, H E L L O ! ! !) in new stack -- Executing [039988120@from-external:2] DumpChan(PJSIP/sipp-, ) in new stack Dumping Info For Channel: PJSIP/sipp-: Info: Name= PJSIP/sipp- Type= PJSIP UniqueID= 172.16.60.160-1394542052.0 LinkedID= 172.16.60.160-1394542052.0 CallerIDNum=39937841;cpc=payphone CallerIDName= 39937841 39937841 ConnectedLineIDNum= (N/A) ConnectedLineIDName=(N/A) DNIDDigits= (N/A) RDNIS= (N/A) Parkinglot= Language= en State= Ring (4) Rings= 1 NativeFormat= (alaw) WriteFormat=alaw ReadFormat= alaw RawWriteFormat= alaw RawReadFormat= alaw WriteTranscode= No ReadTranscode= No 1stFileDescriptor= -1 Framesin= 0 Framesout= 0 TimetoHangup= 0 ElapsedTime=0h0m0s BridgeID= (Not bridged) Context=from-external Extension= 039988120 Priority= 2 CallGroup= PickupGroup= Application=DumpChan Data= (Empty) Blocking_in=(Not Blocking) Variables: -- Executing [039988120@from-external:3] Answer(PJSIP/sipp-, ) in new stack --- Transmitting SIP response (1060 bytes) to UDP:10.25.153.150:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.25.153.150:5060 ;rport;received=10.25.153.150;branch=z9hG4bK587.67258295.0 Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuqC93X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJ3lqwWxarW.gqWReJMEPA36juW6WBzR363RVA3Ejugx3* Record-Route: sip:10.25.153.150;lr;ftag=02e3a8c0-33807b-t-2 Call-ID: 2915b6e4-02e3a8c0-be53@192.168.225.2 From: 39937841 39937841 sip:39937841;cpc=payphone@192.168.225.2 ;user=phone;tag=02e3a8c0-33807b-t-2 To: sip:D39539988120@192.168.225.2 ;user=phone;tag=b23cda89-931c-4a95-85c5-0ec8b03f895c CSeq: 2 INVITE Contact: sip:172.16.60.160:5060 Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 193 v=0 o=- 62708 4 IN IP4 172.16.60.160 s=Asterisk c=IN IP4 172.16.60.160 t=0 0 m=audio 13644 RTP/AVP 8 c=IN IP4 172.16.60.160 a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:150 a=sendrecv --- Received SIP request (703 bytes) from UDP:10.25.153.150:5060 --- ACK sip:172.16.60.160:5060 SIP/2.0 Via: SIP/2.0/UDP 10.25.153.150:5060;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP
[asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint
Hello everyone, I have started testing the PJSIP stack. I saw that it is possible to setup statically multiple AOR contacts, setup qualify_timeout and attach it to an endpoint, and then dial using this endpoint. When I setup the configuration I used the cli in order to see the status of the contacts, and it worked fine - whenever a contact is unreachable, the status is updated to Unavailable. However, when I dial through this endpoint the asterisk doesn't use other contacts which are available in this endpoint. Is it a known issue? Are you planning to solve it? Below is my pjsip.conf: [transport-udp] type=transport protocol=udp bind=172.16.60.160:5060 ;SIPP [sipp] type=endpoint transport=transport-udp context=from-external disallow=all allow=alaw 100rel=required aors=sipp [sipp] type=aor contact=sip:172.16.60.160:5080 contact=sip:10.25.153.150:5060 qualify_frequency=10 [sipp] type=identify endpoint=sipp match=10.25.153.150 match=172.16.60.160 -- _ -- 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] PJSIP - Using multiple AOR contacts when dialing through an endpoint
Yaron Nachum wrote: Hello everyone, I have started testing the PJSIP stack. I saw that it is possible to setup statically multiple AOR contacts, setup qualify_timeout and attach it to an endpoint, and then dial using this endpoint. When I setup the configuration I used the cli in order to see the status of the contacts, and it worked fine - whenever a contact is unreachable, the status is updated to Unavailable. However, when I dial through this endpoint the asterisk doesn't use other contacts which are available in this endpoint. Is it a known issue? Are you planning to solve it? Due to limitations within the Asterisk core you have to use the PJSIP_DIAL_CONTACTS dialplan function[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_DIAL_CONTACTS -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- 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] PJSIP - Using multiple AOR contacts when dialing through an endpoint
Thanks Joshua, I tried it already. That would generate a call to both AORs which is not what I was looking for. Isn't there a way to retrieve the AOR status from the dialplan? On Tue, Mar 11, 2014 at 3:27 PM, Joshua Colp jc...@digium.com wrote: Yaron Nachum wrote: Hello everyone, I have started testing the PJSIP stack. I saw that it is possible to setup statically multiple AOR contacts, setup qualify_timeout and attach it to an endpoint, and then dial using this endpoint. When I setup the configuration I used the cli in order to see the status of the contacts, and it worked fine - whenever a contact is unreachable, the status is updated to Unavailable. However, when I dial through this endpoint the asterisk doesn't use other contacts which are available in this endpoint. Is it a known issue? Are you planning to solve it? Due to limitations within the Asterisk core you have to use the PJSIP_DIAL_CONTACTS dialplan function[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ Function_PJSIP_DIAL_CONTACTS -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- 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
Re: [asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint
Yaron Nachum wrote: Thanks Joshua, I tried it already. That would generate a call to both AORs which is not what I was looking for. Isn't there a way to retrieve the AOR status from the dialplan? Not currently. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- 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] PJSIP - Using multiple AOR contacts when dialing through an endpoint
Thanks for the response anyway. I think that it would be great if someone would make it happen. It seems to me trivial that once you enable to setup multiple AORs you would use them :-) Yaron. On Tue, Mar 11, 2014 at 3:38 PM, Joshua Colp jc...@digium.com wrote: Yaron Nachum wrote: Thanks Joshua, I tried it already. That would generate a call to both AORs which is not what I was looking for. Isn't there a way to retrieve the AOR status from the dialplan? Not currently. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- 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
Re: [asterisk-users] Linux call router
On 11-03-14 12:15, binary dreamer wrote: hello there, I am facing an issue with misd/misdnuser/lcr in the system I am running debian 7 and I managed to install from git misdn/misdnuser but in lcr I am getting: chan_lcr.c: In function 'load_module': chan_lcr.c:3520:24: warning: assignment makes pointer from integer without a cast [enabled by default] make[2]: *** [chan_lcr.po] Error 1 make[2]: Leaving directory /usr/src/lcr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory /usr/src/lcr' make: *** [all] Error 2 root@voyage:/usr/src/lcr# could someone help me please? I have not built lcr in a while (on CentOS) but iirc: make sure you get from http://misdn.eu/ the latest isdn4k-utils, mISDN, mISDNuser and lcr. Then build in that order. For lcr I had these build requirements: autoconf automake libtool libtiff-devel mISDNuser-devel ncurses-devel openssl-devel If you still can't figure it out perhaps ask on the ISDN4Linux mailing list: https://www.isdn4linux.de/mailman/listinfo/isdn4linux Cheers, Patrick ps don't build as root, it's bad practice. -- _ -- 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] Asterisk Authentication
I am not sure if you understood the problem. Asterisk introduced match_auth_username option for what I exactly want but it doesn't seem to be work. On Tue, Mar 11, 2014 at 5:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote: On 11 March 2014 11:39, Jim Boykin boykin...@gmail.com wrote: Hi, I am trying to setup asterisk so that anyone from any IP can call using any callerid as long they have an account - also no registration is required. However, it seems like asterisk tries to find peer based on either the IP address or from header. What I really want is asterisk to find account/peer based on username passed as part of the authentication and NOT from the IP address or the from header. Any idea how to achieve this. Thanks It has to be either fixed IP address or username and password with a dynamic host. This is no in between to the best of my knowledge. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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
Re: [asterisk-users] Asterisk Authentication
Try setting the sip.conf entry to friend, not peer and not user. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Boykin Sent: Tuesday, March 11, 2014 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Authentication I am not sure if you understood the problem. Asterisk introduced match_auth_username option for what I exactly want but it doesn't seem to be work. On Tue, Mar 11, 2014 at 5:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote: On 11 March 2014 11:39, Jim Boykin boykin...@gmail.com wrote: Hi, I am trying to setup asterisk so that anyone from any IP can call using any callerid as long they have an account - also no registration is required. However, it seems like asterisk tries to find peer based on either the IP address or from header. What I really want is asterisk to find account/peer based on username passed as part of the authentication and NOT from the IP address or the from header. Any idea how to achieve this. Thanks It has to be either fixed IP address or username and password with a dynamic host. This is no in between to the best of my knowledge. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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
Re: [asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint
On Tue, Mar 11, 2014 at 8:45 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Thanks for the response anyway. I think that it would be great if someone would make it happen. It seems to me trivial that once you enable to setup multiple AORs you would use them :-) Yaron. On Tue, Mar 11, 2014 at 3:38 PM, Joshua Colp jc...@digium.com wrote: Yaron Nachum wrote: Thanks Joshua, I tried it already. That would generate a call to both AORs which is not what I was looking for. Isn't there a way to retrieve the AOR status from the dialplan? Not currently. We're still adding dialplan functions and CLI commands to the PJSIP stack. Right now there's a way to drill down into endpoint configuration via the PJSIP_ENDPOINT function, but we haven't yet expanded that to AORs. Doing so is a pretty natural next step. There's some discussion of this on the following JIRA issue, where Josh mentions we could query down into the contacts for some of the information: https://issues.asterisk.org/jira/browse/ASTERISK-23173 We'd probably have something similar to PJSIP_ENDPOINT, such as PJSIP_AOR or PJSIP_CONTACT (or something like that), that lets you get at the run-time information of an AOR. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833
On Tue, Mar 11, 2014 at 8:23 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hello, I have installed the latest version 12 that has been released (12.1.0.rc3). I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working. Is it a known issue? I don't think that's an issue at all. Your configured your endpoint to support RFC 4733 DTMF. However, the INVITE request that was received by Asterisk didn't offer support for DTMF, so Asterisk can't accept it. It has to accept only what is in the offer. Your configuration can't force the UA to offer what it wants - you can only configure Asterisk with what it should support with that UA. There's really only two possible outcomes here: (1) Reject the INVITE request with a 488 (you didn't offer me DTMF!) (2) Accept the INVITE request but not have DTMF over RFC 4733. What you're seeing is option (2), which I think is better than rejecting the entire call simply because the thing you are talking to doesn't support the DTMF mode you configured it to have. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Oddity with FFA
Steve, I appreciate you elaborating on my problem. I don't suppose this is as easy as putting a wait(3) in my dial plan before hangup.? (Didn't think so.) Aside from checking (and hoping) for a newer version of FFA that fixes this issue, I guess there's not much I can do, then. Thanks again. Mike. On Tue, Mar 11, 2014 at 12:27 AM, Steve Underwood ste...@coppice.orgwrote: Hi Mike, If the sending machine keeps trying it might be the call has been hung up by asterisk before its own acknowledgement message has finished being sent. There have been bugs like this in the past, and people can be pretty casual about making changes which hang up aggressively. A FAX system should really wait for the final DCN message before disconnecting, to ensure both sides have seen what they need. Spandsp does that, but I am not sure about FFA. Regards, Steve On 03/11/2014 03:03 AM, Mike Diehl wrote: Steve, I BELIEVE the fax is complete because the fax image is a form that appears to only be a single page. But, since FFA isn't providing acknowledgement, the sending fax machine is resending the document multiple times! Mike. On Mon, Mar 10, 2014 at 12:49 PM, Steve Underwood ste...@coppice.orgmailto: ste...@coppice.org wrote: On 03/11/2014 12:36 AM, Mike Diehl wrote: Hi all, For the most part, we are finding that Fax for Asterisk works pretty well. However, we have seen some wierdness that we'd like to try to fix. Once in a while, we will get a partial result report for a given fax but when we look at the actual .tiff image, it looks to be complete. This is causing our users to not get a positive acknowledgement when they send the fax. Is there anything we can do to mitigate this? Mike. How do you know the FAX is complete? If a page was received, the sending machine said more pages were to follow, and then it dropped the call, is that a complete FAX? Steve -- _ -- 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 -- _ -- 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] Asterisk Authentication
actually alwaysauthreject was the problem, making it yes was a solution. thanks everyone On Tue, Mar 11, 2014 at 8:17 PM, Eric Wieling ewiel...@nyigc.com wrote: Try setting the sip.conf entry to friend, not peer and not user. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Boykin Sent: Tuesday, March 11, 2014 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Authentication I am not sure if you understood the problem. Asterisk introduced match_auth_username option for what I exactly want but it doesn't seem to be work. On Tue, Mar 11, 2014 at 5:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote: On 11 March 2014 11:39, Jim Boykin boykin...@gmail.com wrote: Hi, I am trying to setup asterisk so that anyone from any IP can call using any callerid as long they have an account - also no registration is required. However, it seems like asterisk tries to find peer based on either the IP address or from header. What I really want is asterisk to find account/peer based on username passed as part of the authentication and NOT from the IP address or the from header. Any idea how to achieve this. Thanks It has to be either fixed IP address or username and password with a dynamic host. This is no in between to the best of my knowledge. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 -- _ -- 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] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833
Hi Mathew, The regular sip stack has 'auto' dtmfmode which behaved as I said - if the remote replied with telephony event it used RFC2833 otherwise it used inband. On Tue, Mar 11, 2014 at 5:43 PM, Matthew Jordan mjor...@digium.com wrote: On Tue, Mar 11, 2014 at 8:23 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hello, I have installed the latest version 12 that has been released (12.1.0.rc3). I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working. Is it a known issue? I don't think that's an issue at all. Your configured your endpoint to support RFC 4733 DTMF. However, the INVITE request that was received by Asterisk didn't offer support for DTMF, so Asterisk can't accept it. It has to accept only what is in the offer. Your configuration can't force the UA to offer what it wants - you can only configure Asterisk with what it should support with that UA. There's really only two possible outcomes here: (1) Reject the INVITE request with a 488 (you didn't offer me DTMF!) (2) Accept the INVITE request but not have DTMF over RFC 4733. What you're seeing is option (2), which I think is better than rejecting the entire call simply because the thing you are talking to doesn't support the DTMF mode you configured it to have. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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
Re: [asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833
On Tue, Mar 11, 2014 at 11:23 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hi Mathew, The regular sip stack has 'auto' dtmfmode which behaved as I said - if the remote replied with telephony event it used RFC2833 otherwise it used inband. Correct. There is no setting for dtmf_mode that is analogous to the chan_sip 'auto' setting - what you configure for you endpoint today is what it will use. That's not a bug, just something not existing yet. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833
Mathew, Thanks Mathew. It's good to know the limitations :-) Is there any plan to add it? On Tue, Mar 11, 2014 at 6:38 PM, Matthew Jordan mjor...@digium.com wrote: On Tue, Mar 11, 2014 at 11:23 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hi Mathew, The regular sip stack has 'auto' dtmfmode which behaved as I said - if the remote replied with telephony event it used RFC2833 otherwise it used inband. Correct. There is no setting for dtmf_mode that is analogous to the chan_sip 'auto' setting - what you configure for you endpoint today is what it will use. That's not a bug, just something not existing yet. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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
Re: [asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint
Thanks Mathew, That would be great - just to validate the status of the AOR before you send the INVITE. Great mailing list. On Tue, Mar 11, 2014 at 5:38 PM, Matthew Jordan mjor...@digium.com wrote: On Tue, Mar 11, 2014 at 8:45 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Thanks for the response anyway. I think that it would be great if someone would make it happen. It seems to me trivial that once you enable to setup multiple AORs you would use them :-) Yaron. On Tue, Mar 11, 2014 at 3:38 PM, Joshua Colp jc...@digium.com wrote: Yaron Nachum wrote: Thanks Joshua, I tried it already. That would generate a call to both AORs which is not what I was looking for. Isn't there a way to retrieve the AOR status from the dialplan? Not currently. We're still adding dialplan functions and CLI commands to the PJSIP stack. Right now there's a way to drill down into endpoint configuration via the PJSIP_ENDPOINT function, but we haven't yet expanded that to AORs. Doing so is a pretty natural next step. There's some discussion of this on the following JIRA issue, where Josh mentions we could query down into the contacts for some of the information: https://issues.asterisk.org/jira/browse/ASTERISK-23173 We'd probably have something similar to PJSIP_ENDPOINT, such as PJSIP_AOR or PJSIP_CONTACT (or something like that), that lets you get at the run-time information of an AOR. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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
Re: [asterisk-users] Pass Sound files as Argument to Macro Asterisk 1.8
It should be.
I'd write something like the below:
[macro-test]
exten = s,1,NoOp
exten = s,n,GotoIf($[${STAT(e,/var/lib/asterisk/sounds/${ARG1}.ulaw)} =
0]?NOPROMPT:PLAYBACK)
exten = s,n(NOPROMPT),Background(nothing-recordedforpm-prompt-number)
exten = s,n,SayPhonetic(${ARG1})
exten = s,n,Goto(EXIT)
exten = s,n(PLAYBACK),NoOP
exten = s,n,BACKGROUND(${ARG1})
exten = s,n,(EXIT)MacroExit
exten = 1234,1,Macro(macro-test,tt-monkeys)
On Sat, Mar 8, 2014 at 11:39 AM, Daniel van den Berg aster...@suretel.co.za
wrote:
Hi All,
I was wondering if it is possible to pass sound files to a macro as an
argument in Asterisk 1.8?
Thanks!
Regards,
Daniel van den Berg
SureTel
South Africa
087-944-7873
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