Re: [asterisk-users] packages.digium.com
On 11 Mar 2015, at 17:53, Matthew Jordan mjor...@digium.com wrote: On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes steve-li...@geekinter.net wrote: Anyone know where it’s gone?.. Appears to have been down all day. The hamsters should be running in their wheels again now. Cheers Matthew. Give them some food from me. 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
Re: [asterisk-users] switching from SIP to Skype..or not
On Thu, 12 Mar 2015 12:52:46 +, Thufir wrote: Heh, well, I guess it's dead: http://www.digium.com/en/products/software/skype-for-asterisk is this current? http://www.remsys.com/blog/skype-connect-to-asterisk it doesn't solve, I think, the problem I have that SIP clients, sans Asterisk, cannot connect out due to too many hops/bad connection. Only Skype is able, from home at least, to connect out. From what I can tell. -Thufir -- _ -- 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] switching from SIP to Skype..or not
Your characterization may be true but Skype works much better than SIP when it comes to sound quality. I have SIP softphone with Asterisk server and Skype on the same workstation. Skype just works better over the same network. Ron On 12/03/2015 9:26 AM, A J Stiles wrote: On Thursday 12 Mar 2015, Thufir wrote: I'm testing Asterisk at home, crummy connection. Skype works fine for me, but every SIP client, even without using Asterisk, fails to connect. That's ok. Is swapping out SIP for Skype a big deal? Stay away from Skype! It is a toxic, proprietary product. The lack of interoperability by design is the antithesis of what a telecommunication system should be about -- and the extent to which they have gone to thwart any attempt at interoperability is truly shocking. For connecting two Asterisk installations to each other over the Internet, IAX is better than SIP -- that's what it was designed for. -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] switching from SIP to Skype..or not
Hey all We have been working with SIP for years. It has the potential to be better than Skype. It is really all in the implementation. Not all SIP soft clients are equal nor are the networks and computers they are running on. I will not bash Skype. We have tested it and in most cases choose not to use it. It has it's place and is good for the user that meets it's specific target demographic. SIP is a sold communications protocol that can communication with codecs of differ audio and video quality levels, and supports industry standard software and hardware endpoints. With SIP you get to choose how good your quality is. With Skype Microsoft does. It comes down to what do you want to achieve, how much resource do you want to put in to it, and are you committed to a bit more work for a lot more options and better quality, or do you want a quick and easy solution with differing limits. Both solutions have their place. To me SIP vs Skype is like complaining apples and carrots do you want fruit or veggies you get to choose. You can choose to agree or disagree with my statements. I hope they are useful to some. Thanks Bryant From: Ron Wheeler rwhee...@artifact-software.com Sent: Thursday, March 12, 2015 9:40 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] switching from SIP to Skype..or not Your characterization may be true but Skype works much better than SIP when it comes to sound quality. I have SIP softphone with Asterisk server and Skype on the same workstation. Skype just works better over the same network. Ron On 12/03/2015 9:26 AM, A J Stiles wrote: On Thursday 12 Mar 2015, Thufir wrote: I'm testing Asterisk at home, crummy connection. Skype works fine for me, but every SIP client, even without using Asterisk, fails to connect. That's ok. Is swapping out SIP for Skype a big deal? Stay away from Skype! It is a toxic, proprietary product. The lack of interoperability by design is the antithesis of what a telecommunication system should be about -- and the extent to which they have gone to thwart any attempt at interoperability is truly shocking. For connecting two Asterisk installations to each other over the Internet, IAX is better than SIP -- that's what it was designed for. -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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 13.2.0 Video issues
On Tue, Mar 10, 2015 at 5:22 PM, Toufic Khreish (Gmail) toufic.khre...@gmail.com wrote: Thank you, I needed a starting point to start my post. 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. Voice issues on IAX2 Trunks, All extensions are SIP. The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : iax2 set debug trunk on [2015-03-10 16:28:42] WARNING[9614][C-000b]: chan_iax2.c:1793 compress_subclass: Can't compress subclass 2097217 On the box running asterisk 1.6.2.6 I receive the following warning: [2015-03-10 16:35:00] WARNING[24872]: translate.c:168 framein: no samples for alawtolin core show channels Channel Location State Application(Data) IAX2/Mypbx1-15288(None) Up AppDial((Outgoing Line)) SIP/6000-000f(None) Up Dial(IAX2/Mypbx1/300,300,Tt) 2 active channels Trunks are between an asterisk 1.6.2.6 and asterisk 12.8.1 (IAX , Alaw and GSM codecs) Voice is not very clear and choppy If I try the same between an asterisk 13.2.0 and the asterisk 1.6.2.6 , voice is very clear. Both Asterisk 1.6.2 and Asterisk 12 no longer receive bug fixes, so I'm going to skip past this issue. 2. Asterisk 13.2.0 Video issues (no IAX2 voice issues). Calls from Bria video sip phone (android or IOS) to Grandstream GXV3175 (asterisk engine stops/crashes) Asterisk crashing is a bug. That's a bad thing. Please get a backtrace [1] and file an issue on the issue tracker [2]. A pcap of the message traffic would also be very helpful. Call from Groundwire video sip (IOS since Android version does not H264 codec) to Grandstream GXV3175, Asterisk stops I'm going to assume Asterisk stops means it crashed as well. If you'd like to get a backtrace for that as well and attach it to the same issue, that would be helpful - it may be the same problem that you see with the Bria phone, or it may be something else. Calls between SIP Video softphones works well no issues. Well, that's good. :-) Calls from SIP video softphones (BRIA) to Grandstream GXV3275 works well. (Acrobitz Groundwire to GXV3275 picture on Grandstream is yellowish) Calls between GXV3275 and GXV3175 video streaming is very slow on the GXV3175 (this is not the case under Asterisk 12.8.1) Calls from GXV3175 to Bria (video is displayed on bria side only) Since there are some that work fine, and some that don't, the trick is going to be knowing: (1) How the SIP peers (or PJSIP endpoints) are configured (2) How the phones are negotiating media with Asterisk Both your SIP configuration as well as a DEBUG log - generated with trace logging, showing the negotiation [3] - will be needed to figure out what is occurring. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira/ [3] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Matthew Jordan Digium, Inc. | Director of Technology 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 and Kamailio without registration
On Tue, Mar 10, 2015 at 6:11 PM, Chirag Desai djchill...@gmail.com wrote: OK, it stopped working. It turns out the transport and endpoints in PJSIP are ok. I can send an invite from my unregistered snom phone and I can see some activity in the CLI. However, when I dial from my snom to Kamailio and have it pass the message to asterisk, PJSIP seems to ignore the sip messages even though they are there. Is there something wrong in the invite that I'm missing? U 2015/03/10 22:47:43.539208 [kamailio public ip]:5060 - [asterisk public ip]:5061 INVITE sip:1...@somedomain.com;user=phone SIP/2.0. Record-Route: sip:[kamailio public ip];r2=on;lr=on;nat=yes. Record-Route: sip:[kamailio public ip];transport=tcp;r2=on;lr=on;nat=yes. Via: SIP/2.0/UDP 1 [kamailio public ip];branch=z9hG4bKc10a.a307d27e5d7581c259704fcd865a69e2.0;i=1. Via: SIP/2.0/TCP [snomprivateip]:47153;received=[snompublicip];branch=z9hG4bK-cc9ldmdhvvdi;rport=47473. From: sip:1...@somedomain.com;tag=tu0if9akzq. To: sip:451...@somedomain.com;user=phone. Call-ID: 8d74ff54e076-hajfjxwp1crj. CSeq: 2 INVITE. Max-Forwards: 16. Contact: sip:1000@[snom_private_ip]:47153;alias=[snom_public_ip]~47473~2;transport=tcp;line=snl8cukk;reg-id=1. X-Serialnumber: [snom_mac_address]. P-Key-Flags: resolution=31x13, keys=4. User-Agent: snom760/8.7.3.25. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE. Allow-Events: talk, hold, refer, call-info. Supported: timer, 100rel, replaces, from-change. Session-Expires: 3600;refresher=uas. Min-SE: 90. Content-Type: application/sdp. Content-Length: 598. . v=0. o=root 1667335791 1667335791 IN IP4 [snom_private_ip]. s=call. c=IN IP4 [snom_private_ip]. t=0 0. m=audio 59358 RTP/AVP 9 0 8 3 97 98 99 100 106 107 108 109 18 101. a=rtpmap:9 G722/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:97 G726-16/8000. a=rtpmap:98 G726-24/8000. a=rtpmap:99 G726-32/8000. a=rtpmap:100 G726 My transports are: [transport-udp] type=transport protocol=udp bind:0.0.0.0:5061 [transport-tcp] type=transport protocol=tcp bind=0.0.0.0:5061 If the INVITE request is not shown in the CLI with 'pjsip set logger on', then Asterisk is not actually receiving the request. Does a pcap show the message being sent to the correct IP/port? If you change the transports to bind to port 5060, does that change anything? -- Matthew Jordan Digium, Inc. | Director of Technology 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] switching from SIP to Skype..or not
On Thursday 12 Mar 2015, Thufir wrote: I'm testing Asterisk at home, crummy connection. Skype works fine for me, but every SIP client, even without using Asterisk, fails to connect. That's ok. Is swapping out SIP for Skype a big deal? Stay away from Skype! It is a toxic, proprietary product. The lack of interoperability by design is the antithesis of what a telecommunication system should be about -- and the extent to which they have gone to thwart any attempt at interoperability is truly shocking. For connecting two Asterisk installations to each other over the Internet, IAX is better than SIP -- that's what it was designed for. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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 some AMI events is absent?
On Wed, Mar 11, 2015 at 7:11 PM, Jean-Denis Girard jd.gir...@sysnux.pf wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I made some tests with asterisk-13.2.0 and chan_pjsip this weekend myself, and came to the same conclusion: some peerstatus events are missing (eg. when contacts become unreachable / unavailable, IIRC), and I could not find a way to get contacts status through AMI. It looks a bit similar to issues 23172, 23173: PJSip missing functionalities. Yup, it was an oversight in the implementation of 'qualify' for PJSIP endpoints. Dmitriy was kind enough to open an issue for it: https://issues.asterisk.org/jira/browse/ASTERISK-24863 -- Matthew Jordan Digium, Inc. | Director of Technology 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] switching from SIP to Skype..or not
On 3/12/15 9:39 AM, Ron Wheeler wrote: Your characterization may be true but Skype works much better than SIP when it comes to sound quality. SIP is not to blame for this. Its the audio codec being used. Skype has spend a great deal of effort with their SILK codec by making it highly tolerant of packet loss and jitter. The same cannot be said for the standard codecs Asterisk uses. I have SIP softphone with Asterisk server and Skype on the same workstation. Skype just works better over the same network. Ron On 12/03/2015 9:26 AM, A J Stiles wrote: On Thursday 12 Mar 2015, Thufir wrote: I'm testing Asterisk at home, crummy connection. Skype works fine for me, but every SIP client, even without using Asterisk, fails to connect. That's ok. Is swapping out SIP for Skype a big deal? Stay away from Skype! It is a toxic, proprietary product. The lack of interoperability by design is the antithesis of what a telecommunication system should be about -- and the extent to which they have gone to thwart any attempt at interoperability is truly shocking. For connecting two Asterisk installations to each other over the Internet, IAX is better than SIP -- that's what it was designed for. -- Technical Support http://www.cellroute.net -- _ -- 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] switching from SIP to Skype..or not
Which wideband codec did you use when testing SIP? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler Sent: Thursday, March 12, 2015 9:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] switching from SIP to Skype..or not Your characterization may be true but Skype works much better than SIP when it comes to sound quality. I have SIP softphone with Asterisk server and Skype on the same workstation. Skype just works better over the same network. Ron -- _ -- 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] switching from SIP to Skype..or not
On Thu, Mar 12, 2015 at 10:04:08AM -0400, Andres wrote: On 3/12/15 9:39 AM, Ron Wheeler wrote: Your characterization may be true but Skype works much better than SIP when it comes to sound quality. SIP is not to blame for this. Its the audio codec being used. Skype has spend a great deal of effort with their SILK codec by making it highly tolerant of packet loss and jitter. The same cannot be said for the standard codecs Asterisk uses. Opus was co-developed by Skype and could be used with Asterisk (if support to it was added). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- 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] chanspy for group extension
On 3/12/15 12:19 PM, Administrator TOOTAI wrote:
Hi,
Le 12/03/2015 17:28, Salaheddine Elharit a écrit :
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},__dqs)
Here you have a problem: ${EXTEN} value is s
[...]
Daniel
Oops, my bad, that should have been ${MACRO_EXTEN} so it gets the vaule
of the extension that was dialed.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
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Re: [asterisk-users] Unstable phone connection
D'Arcy J.M. Cain If the device is registering and then dropping there are several usual items. The router may be closing the ports on the device. The router may have a AGL SIP helper that is causing issues. Make sure that the device is sending out keep alive packets. Shut down any AGL helpers on the router. Make sure that the site is not double NATing Try using a stun server and see if that helps at all. Watch you console on your sip serer to see how long the device runs before losing connection. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: D'Arcy J.M. Cain da...@vex.net Sent: Thursday, March 12, 2015 2:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unstable phone connection This is driving me to distraction. I have a switch with multiple clients who are all working fine except for one and I can't figure out what makes them different. I have tried every NAT setting in the ATA (SPA112 ATA with 2 x FXS, 1 x LAN), stun server on and off, different sip ports, different RTP ports and it still fails. I have left the location with it working only to have it fail later. He always gets registered but when a call is sent it doesn't respond so the caller hears no ring and the phone does not ring. Yesterday he mentioned that when the phone is working the WiFi slows down significantly. No idea why or if it is related. He has a radio station streaming music. I wondered if that might be interfering. That's why I tried changing the SIP port and the RTP ports but that didn't seem to help. It smells like a network problem to me but I am running the same ADSL device here and other clients are working behind a NAT gateway so I am at a loss as to what might be wrong. Could it be the streaming? Cheers. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- 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] GXP 1405 and asterisk
2015-03-12 13:07 GMT-06:00 Bryant Zimmerman brya...@zktech.com: SIPAddHeader(Alert-Info:\;info=ring3) In the phone config add the value ring3 and select Account # / Call Settings / Match Incoming Caller ID (Matching Rule) In the first rule place the word ring3 and select your ring tone. This will cause the selected ringtone to be used when calls with the info value of ring3 is matched can not get it to work any idea o tips? regardss work gr8 , thnk thnk ..Bryant -- rickygm http://gnuforever.homelinux.com -- _ -- 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] chanspy for group extension
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = s,n,Hangup
app-chanspy]
exten = _0071XX,*1,*Macro(chanspy,1234)
exten = _0072XX,*1,*Macro(chanspy,5678)
exten = _0073XX,*1,*Macro(chanspy,8910)
but when i do 007100 for exemple i spy another agnet 102 or 103
any help please
thanks and regards
2015-03-12 10:30 GMT+00:00 Salaheddine Elharit salah.elharit...@gmail.com:
thank you so much it work
you must add 1 like below
[app-chanspy]
exten = _0071XX,*1,*Macro(chanspy,1234)
exten = _0072XX,*1,*Macro(chanspy,5678)
exten = _0073XX,*1,*Macro(chanspy,8910)
best regards.
2015-03-11 19:48 GMT+00:00 Carlos Chavez cur...@telecomabmex.com:
On 3/11/15 12:48 PM, Salaheddine Elharit wrote:
hello list,
i use chanspy with the code below
[app-chanspy]
exten = _007.,1,Macro(user-callerid,)
exten = _007.,n,Answer
exten = _007.,n,Authenticate()
exten = _007.,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = _007.,n,Hangup
i have a question related to chanspy
i have created extension from 100 to 300 and i will give the permission
with group of extension
i want to use chanspy like below
100=199 with Authenticate(1234)
200=299 with Authenticate(5678)
300=399 with Authenticate(8910)
Use a macro and pass the pin as a parameter:
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = s,n,Hangup
[app-chanspy]
exten = _0071XX,Macro(chanspy,1234)
exten = _0072XX,Macro(chanspy,5678)
exten = _0073XX,Macro(chanspy,8910)
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] packages.digium.com
On Thu, 12 Mar 2015 08:58:01 + Steven Howes steve-li...@geekinter.net wrote: On 11 Mar 2015, at 17:53, Matthew Jordan mjor...@digium.com wrote: On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes steve-li...@geekinter.net wrote: Anyone know where it’s gone?.. Appears to have been down all day. The hamsters should be running in their wheels again now. Cheers Matthew. Give them some food from me. But remember, the key to an HA hamster cluster is staggered feeding times! -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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] Unstable phone connection
On Thu, 12 Mar 2015 15:14:24 -0400 Bryant Zimmerman brya...@zktech.com wrote: If the device is registering and then dropping there are several usual items. The router may be closing the ports on the device. I don't see how. I am logged into the ATA through the router and I don't lose the connection. The router may have a AGL SIP helper that is causing issues. Can't find an AGL setting. There is a SIP checkbox. Pretty sure I have that turned off but I can try to get someone to check. Make sure that the device is sending out keep alive packets. I have that flag turned on. Shut down any AGL helpers on the router. See above. Make sure that the site is not double NATing There's only one router. It is the ADSL device as well. Try using a stun server and see if that helps at all. I tried with and without. I am using stunserver.org. Watch you console on your sip serer to see how long the device runs before losing connection. I don't think it does. Both the server and the ATA think that they are still registered but when a call comes in there is no ringing on the line. If I split dial it rings the cell phone but I still hear no ringing from the caller side unless registration is actually turned off from the ATA and a sip unregister is issued. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- 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] Unstable phone connection
This is driving me to distraction. I have a switch with multiple clients who are all working fine except for one and I can't figure out what makes them different. I have tried every NAT setting in the ATA (SPA112 ATA with 2 x FXS, 1 x LAN), stun server on and off, different sip ports, different RTP ports and it still fails. I have left the location with it working only to have it fail later. He always gets registered but when a call is sent it doesn't respond so the caller hears no ring and the phone does not ring. Yesterday he mentioned that when the phone is working the WiFi slows down significantly. No idea why or if it is related. He has a radio station streaming music. I wondered if that might be interfering. That's why I tried changing the SIP port and the RTP ports but that didn't seem to help. It smells like a network problem to me but I am running the same ADSL device here and other clients are working behind a NAT gateway so I am at a loss as to what might be wrong. Could it be the streaming? Cheers. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- 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] chanspy for group extension
Hi,
Le 12/03/2015 17:28, Salaheddine Elharit a écrit :
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},__dqs)
Here you have a problem: ${EXTEN} value is s
[...]
Daniel
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[asterisk-users] GXP 1405 and asterisk
Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten = 0,1,Playback(pls-wait-connect-call) same= n,SIPAddHeader(Alert-Info:;info=ring3) same= n,Dial(SIP/310SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chanspy for group extension
This is one of the drawbacks to using macros. There are workarounds for
macros, but the correct solution is use the Gosub / Return dialplan applications
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Thursday, March 12, 2015 2:19 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] chanspy for group extension
Hi,
Le 12/03/2015 17:28, Salaheddine Elharit a écrit :
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},__dqs)
Here you have a problem: ${EXTEN} value is s
[...]
Daniel
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Re: [asterisk-users] GXP 1405 and asterisk
SIPAddHeader(Alert-Info:\;info=ring3) In the phone config add the value ring3 and select Account # / Call Settings / Match Incoming Caller ID (Matching Rule) In the first rule place the word ring3 and select your ring tone. This will cause the selected ringtone to be used when calls with the info value of ring3 is matched Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: ricky gutierrez xserverli...@gmail.com Sent: Thursday, March 12, 2015 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] GXP 1405 and asterisk Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten = 0,1,Playback(pls-wait-connect-call) same= n,SIPAddHeader(Alert-Info:;info=ring3) same= n,Dial(SIP/310SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chanspy for group extension
thank you but could you please tell me how can i put it
thanks and regards
2015-03-12 18:19 GMT+00:00 Administrator TOOTAI ad...@tootai.net:
Hi,
Le 12/03/2015 17:28, Salaheddine Elharit a écrit :
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},__dqs)
Here you have a problem: ${EXTEN} value is s
[...]
Daniel
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[asterisk-users] switching from SIP to Skype..or not
I'm testing Asterisk at home, crummy connection. Skype works fine for me, but every SIP client, even without using Asterisk, fails to connect. That's ok. Is swapping out SIP for Skype a big deal? Heh, well, I guess it's dead: http://www.digium.com/en/products/software/skype-for-asterisk If I have a really bad connection, can I downgrade SIP somehow? I don't really need to use to make voice calls. Or, more specifically, quality, echo, distortion aren't relevant. Just SIP to SIP hello. When I connect to any SIP provider, ekiga, etc, without using Asterisk, I get too many hops errors. While I have another computer on the LAN I can connect to, it's not quite the same. Any thoughts? thanks, Thufir -- _ -- 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 and Kamailio without registration
On Thu, Mar 12, 2015 at 5:11 PM, Chirag Desai djchill...@gmail.com wrote: From: Matthew Jordan mjor...@digium.com If the INVITE request is not shown in the CLI with 'pjsip set logger on', then Asterisk is not actually receiving the request. Does a pcap show the message being sent to the correct IP/port? If you change the transports to bind to port 5060, does that change anything? The sip message I included in my last message is what I see when I ngrep on 5061, but asterisk doesn't see it. When I tell Kamailio to send the message to 5060 chan_sip shows the invite in the CLI. My setup has chan_sip running on 5060 and pjsip (tcp and udp on 5061). I'll get PJSIP running on 5060 and see if that makes any difference. UPDATE: I got PJSIP on 5060 and everything is working fine as expected and I can see the calls from Kamalio. Is this a bug with asterisk not recognising the traffic on 5061 even though the SIP messages are being received by the server on that port and I can see it? I suspect not. We run the PJSIP stack on multiple ports quite often. I would guess that there's something else going on here. -- Matthew Jordan Digium, Inc. | Director of Technology 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
[asterisk-users] Realtime followme and channel variables
Followme is perfect to handle FMFM and it is now also realtime, but it seems impossible to assign some value to a variable, from within the followme to store info for example about the tenant the followme is running under, like instead happen for example in the queue with the setinterfacevar field. I just need to pass a variable from the channel placing the call to the followme to the channel where the extension is dialed by followme. Any idea? Leandro -- _ -- 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 and Kamailio without registration
From: Matthew Jordan mjor...@digium.com If the INVITE request is not shown in the CLI with 'pjsip set logger on', then Asterisk is not actually receiving the request. Does a pcap show the message being sent to the correct IP/port? If you change the transports to bind to port 5060, does that change anything? The sip message I included in my last message is what I see when I ngrep on 5061, but asterisk doesn't see it. When I tell Kamailio to send the message to 5060 chan_sip shows the invite in the CLI. My setup has chan_sip running on 5060 and pjsip (tcp and udp on 5061). I'll get PJSIP running on 5060 and see if that makes any difference. UPDATE: I got PJSIP on 5060 and everything is working fine as expected and I can see the calls from Kamalio. Is this a bug with asterisk not recognising the traffic on 5061 even though the SIP messages are being received by the server on that port and I can see it? -- _ -- 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 and Kamailio without registration
From: Matthew Jordan mjor...@digium.com If the INVITE request is not shown in the CLI with 'pjsip set logger on', then Asterisk is not actually receiving the request. Does a pcap show the message being sent to the correct IP/port? If you change the transports to bind to port 5060, does that change anything? The sip message I included in my last message is what I see when I ngrep on 5061, but asterisk doesn't see it. When I tell Kamailio to send the message to 5060 chan_sip shows the invite in the CLI. My setup has chan_sip running on 5060 and pjsip (tcp and udp on 5061). I'll get PJSIP running on 5060 and see if that makes any difference. -- C -- _ -- 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] chanspy for group extension
thank you so much it work
you must add 1 like below
[app-chanspy]
exten = _0071XX,*1,*Macro(chanspy,1234)
exten = _0072XX,*1,*Macro(chanspy,5678)
exten = _0073XX,*1,*Macro(chanspy,8910)
best regards.
2015-03-11 19:48 GMT+00:00 Carlos Chavez cur...@telecomabmex.com:
On 3/11/15 12:48 PM, Salaheddine Elharit wrote:
hello list,
i use chanspy with the code below
[app-chanspy]
exten = _007.,1,Macro(user-callerid,)
exten = _007.,n,Answer
exten = _007.,n,Authenticate()
exten = _007.,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = _007.,n,Hangup
i have a question related to chanspy
i have created extension from 100 to 300 and i will give the permission
with group of extension
i want to use chanspy like below
100=199 with Authenticate(1234)
200=299 with Authenticate(5678)
300=399 with Authenticate(8910)
Use a macro and pass the pin as a parameter:
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = s,n,Hangup
[app-chanspy]
exten = _0071XX,Macro(chanspy,1234)
exten = _0072XX,Macro(chanspy,5678)
exten = _0073XX,Macro(chanspy,8910)
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
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Re: [asterisk-users] Realtime followme and channel variables
On Thu, Mar 12, 2015 at 5:14 PM, Leandro Dardini ldard...@gmail.com wrote: Followme is perfect to handle FMFM and it is now also realtime, but it seems impossible to assign some value to a variable, from within the followme to store info for example about the tenant the followme is running under, like instead happen for example in the queue with the setinterfacevar field. I just need to pass a variable from the channel placing the call to the followme to the channel where the extension is dialed by followme. Any idea? Sounds like you need to use variable inheritance. https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance 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
[asterisk-users] WebRTC demo phones
Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the enable video checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with Rejecting secure video stream without encryption details. - sipML5, but it won't register, perhaps something to do with not using the Asterisk Websocket server (which I don't see an option to choose) - Janus, but the INVITE SDP contains RTP/AVP not RTP/SAVP, and Asterisk rejects it with We are requesting SRTP for audio, but they responded without it! Thanks for any suggestions. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] WebRTC demo phones
Sipml5 works. You need to have TLS enabled on asterisk web socket. Mitul On 12-Mar-2015 12:47 PM, David Cunningham dcunning...@voisonics.com wrote: Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the enable video checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with Rejecting secure video stream without encryption details. - sipML5, but it won't register, perhaps something to do with not using the Asterisk Websocket server (which I don't see an option to choose) - Janus, but the INVITE SDP contains RTP/AVP not RTP/SAVP, and Asterisk rejects it with We are requesting SRTP for audio, but they responded without it! Thanks for any suggestions. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] WebRTC demo phones
Hello David, I'd recommend trying http://sipjs.com/ , it's similar to sipjs but you can choose which kind of media it uses via a configuration object: http://sipjs.com/guides/make-call/ Check out the guides, they are extremely clear and informative: http://sipjs.com/guides/ cheers, Olli 2015-03-12 9:20 GMT+02:00 Mitul Limbani mi...@enterux.in: Sipml5 works. You need to have TLS enabled on asterisk web socket. Mitul On 12-Mar-2015 12:47 PM, David Cunningham dcunning...@voisonics.com wrote: Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the enable video checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with Rejecting secure video stream without encryption details. - sipML5, but it won't register, perhaps something to do with not using the Asterisk Websocket server (which I don't see an option to choose) - Janus, but the INVITE SDP contains RTP/AVP not RTP/SAVP, and Asterisk rejects it with We are requesting SRTP for audio, but they responded without it! Thanks for any suggestions. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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
