Re: [asterisk-users] Recommendation for one chip GSM gateway -- Yeastar vs. Dinstar
Sorry to bump such an old post. Which hub is that? -- _ -- 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 seding 2 INVITEs all of a sudden
Hello Everyone, Today we observed asterisk sending two invites for the initial call before the call was established (ie, not re-invites). There were no changes made to the configuration for a very long time, and was kind of confused when seeing this action. Can someone please suggest where to look to remove this behaviour? U 2014/08/12 07:34:20.405029 192.168.2.10:5060 - 192.168.2.20:5080 INVITE sip:873359633037@192.168.2.20:5080 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK541d5594;rport. Max-Forwards: 70. From: 555955599 sip:555955...@victoria.example.com;tag=as285d2896. To: sip:873359633037@192.168.2.20:5080. Contact: sip:555955599@192.168.2.10:5060. Call-ID: 5a51eef8064a0d360009f64e34c70...@victoria.example.com. CSeq: 102 INVITE. User-Agent: EXAMPLE Systems. Date: Tue, 12 Aug 2014 11:34:20 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 279. . v=0. o=root 1631923320 1631923320 IN IP4 192.168.2.10. s=EXAMPLE Systems. c=IN IP4 192.168.2.10. t=0 0. m=audio 52034 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2014/08/12 07:34:20.903830 192.168.2.10:5060 - 192.168.2.20:5080 INVITE sip:873359633037@192.168.2.20:5080 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK541d5594;rport. Max-Forwards: 70. From: 555955599 sip:555955...@victoria.example.com;tag=as285d2896. To: sip:873359633037@192.168.2.20:5080. Contact: sip:555955599@192.168.2.10:5060. Call-ID: 5a51eef8064a0d360009f64e34c70...@victoria.example.com. CSeq: 102 INVITE. User-Agent: EXAMPLE Systems. Date: Tue, 12 Aug 2014 11:34:20 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 279. . v=0. o=root 1631923320 1631923320 IN IP4 192.168.2.10. s=EXAMPLE Systems. c=IN IP4 192.168.2.10. t=0 0. m=audio 52034 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. Thanks in Advance, Nick -- _ -- 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 seding 2 INVITEs all of a sudden
Thanks Scott, Restarted all the machines since there uptime was 8 years :). Everything works ok now. Kind Regards, Nick. -- _ -- 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] Any way to get rid of X-Asterisk?
Long story... Would be nice if we can remove this on BYEs X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Kind Regards, Nick. -- _ -- 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] Any way to get rid of X-Asterisk?
Yeah I can do that Anything in sip.conf that we can set? N. On Wed, Jul 23, 2014 at 4:39 PM, David Lam software...@gmail.com wrote: This is defined in chan_sip.c. Simply edit the source file and recompile. On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo sym...@gmail.com wrote: Long story... Would be nice if we can remove this on BYEs X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Kind Regards, Nick. -- _ -- 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] Any way to get rid of X-Asterisk?
From sip.conf.sample in 11.10.0 ;use_q850_reason = no ; Default no ; Set to yes add Reason header and use Reason header if it is available. Using 1.8.7. Shiza Thanks as always guys. N. -- _ -- 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] 200 OK however still rinnging
Hello Everyone, We are seeing many instances where we receive 200OK from the vendors however, asterisk still keeps ringing. Is there anyway to stop this from happening? I remember reading something about early media however this seems to be a case of late media? Kind Regards, Nick from Toronto. -- _ -- 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] 200 OK however still rinnging
Did some more testing, what we found was some calls work perfectly to some phone numbers (ie, two way audio). For other phone numbers in the UK we are getting 200 OK however: 1) We hear ringing in the handset 2) Call connected but not audio. This problem is reproducible every time. Our asterisk box is behind nat. Please Help, Nick. -- _ -- 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] Open Source Asterisk Polling Solution
Hello Everyone, Thank you all for your response. The people I am doing it for run a non-profit charity, and are legally able to reach out to their customers. I will wire it up to the DNC however, for starters, I would like to get asterisk to: i) Iterate through a list of numbers ii) Play a pre-recorded message asking if they have waste they need picked up iii) If they press one, forward the call to mailbox The easier the better for us. I did see Wombat, newfies, and vicdial however, I can't go through with the installation process and find out there is some hidden clause, limited to 2 channels etc If I can do it with a simple dialplan as mentioned earlier, I think it's the best solution for starters. Kind Regards, Nick from Toronto -- _ -- 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] Open Source Asterisk Polling Solution
That's about as simple as it gets. A call file that goes to the dialplan. A dialplan that consists of Read (which would play the message) followed a GotoIf into a mailbox (either voicemail or Dial() to an external number). One hint for doing unattended dialing like this, make sure you're dialing using a SIP or other digital method rather than, say, out an analogue port that doesn't have decent answer detect. And you can't just dump a whole bunch of call files into the system at once, you'll need to meter them out based on the number of concurrent outbound calls your provider will allow. Hello James, Good to see you here, and thank you very much. Though my basic idea of how it will look using call files and dialplan is like what you and others on here have pointed out. Yes, we are using SIP for both origination and termination (just helping my friend use some of our accounts used for PBX, and prepaid). I have been using * for many years now however, never for call center/predictive dialer type processes. Once I have got this thing to call out and get calls coming in. It would be nice to write to a database all the users that press option on. I have a strong Java, PHP and SQL background. Will probably need to make a call using AGI or such? N. -- _ -- 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] Open Source Asterisk Polling Solution
Hello Everyone, We are looking for a simple open source auto dialer with polling capabilities. What we would like is a program that we can upload leads to, and have asterisk: i) Dial numbers ii) Play pre-recorded iii) If user presses one, forward the call to an agent There are so many solutions out there it's hard to make a decision on what works, what has just a limited free version etc Something that can support 10 channels, and is stable would be greatly appreciated. If this can be simply implemented using asterisk and call folder, even better PS Our preferred version of * is 1.8.x Kind Regards, Nick from Toronto. -- _ -- 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] Open Source Asterisk Polling Solution
On Mon, Apr 21, 2014 at 2:01 PM, James Sharp ja...@fivecats.org wrote: On 4/21/2014 1:47 PM, Mitul Limbani wrote: Use vicidial for achieving the same. Or call files (or AMI originate), a short bit of dialplan logic, and maybe a call to Queue(). This is a nice and easy solution however, I do not know where to begin. Can you gents kindly elaborate or point us to the right directions (ie, howto tutorials) -- _ -- 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] Open Source Asterisk Polling Solution
I'll be attempting this tomorrow for a friend as a favour. Will post the end result for others in the future. Nick from Toronto. -- _ -- 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] Host = Dynamic in a Register Free Setup
Is Asterisk not able to gather IP and port info on INVITES in a REGISTER free, host=dynamic setup? As you all know REGISTERS are resourceful and the phone can be anywhere.. Kind Regards, Nick. -- _ -- 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] Host = Dynamic in a Register Free Setup
Hello Eric, On 2/18/14, Eric Wieling ewiel...@nyigc.com wrote: No. Asterisk will accept calls from unregistered devices, but you have to enable guests I sip.conf and hope your dialplan is secure. No sane person does this. Thank you for your response. Our security layer is abstracted out of the Asterisk boxes completely. The setup is secure we just need Asterisk to fill in the gaps (ie, ipaddrss and port) after initial INVITE vs. REGISTER. Kind Regards, Nick. -- _ -- 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] Host = Dynamic in a Register Free Setup
I just want to clarify. We are not creating peers automatically. And we allowguest=no. We do have peer entries in sip_buddies db as you would expect. As mentioned, we just don't allow phones to REGISTER every 3600 (for example). Once a valid peer/phone tries to place a call, we would like asterisk to store the ipaddr and port of the phone to be able to create the SIP channel for incoming calls. As mentioned security is managed elsewhere. N. -- _ -- 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] Retaining P-Asserted Info
Hello Ishfaq, I just tried it and it did create a P-Asserted header however it contains the extension of the asterisk peer not what was passed by our switch. From the previous example: P-Asserted-Identity: 222 sip:222@192.168.2.10 (which is asterisk peer extension and not) P-Asserted-Identity: John Doe sip:14167493...@toronto.location.com; user=phone; nat=yes. (which is being passed by the call leg) Is there a flag that retains the rpid from the call leg? Kind Regards, Nick -- _ -- 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] Retaining P-Asserted Info
Hey Guys, I really appreciate this and I apologize for asking however, we do not have any way to test in advance outside of our live environment. Can someone kindly provide a working extension rule that will retain the following P-Asserted info that is existent from the inbound-leg to the outbound-leg using `SIPAddHeader`: P-Asserted-Identity: John Doe sip:14167493...@toronto.location.com; user=phone; nat=yes. Forgive the noob, Nick. -- _ -- 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] Host = Dynamic in a Register Free Setup
Shiza Sounds about right but is it true? Anyone else? -- _ -- 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] Retaining P-Asserted Info
Hello Markus, Thank you so much for your response. Our switch is already generating the needed P-Asserted header: P-Asserted-Identity: John Doe sip:14167493...@toronto.location.com; user=phone; nat=yes. I really did not want to have to rebuild it using `SIPAddHeader` however, if I have no choice, can someone please provide an extension rule that will include the exiting inbound leg line above in the outbound leg. Kind Regards, Nick. -- _ -- 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] Host = Dynamic in a Register Free Setup
Hello Everyone. Our environment is a register free setup, and our phones are set as host=dynamic. The problem we are experiencing is for inbound calls: Name/username HostDyn Forcerport ACL Port Status Realtime 222/222 (Unspecified)D N A 0 Unmonitored Cached RT So when we DIAL 222 we get: WARNING[23103]: app_dial.c:2198 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) My question is how can we get Asterisk to fill in the gaps (ie, ipaddr, port) for a dynamic peer in a register free environment. Kind Regards, Nick. -- _ -- 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] Retaining P-Asserted Info
Hello Everyone, Our switch is sending P-Asserted info to asterisk however the information is getting removed when asterisk forks the call. Is it possible to have asterisk retain the P-Asserted on the leg. This is quite important for valid functionality of our network. Tried setting `sendrpid = yes` and still same problem. We really don't want to have to `SipAddHeader` as it is already being formed by our switch. Thanks in Advance, Nick -- _ -- 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] qualify=yes outboundproxy
We use opensips as a type of firewall as well and don't need to set qualify=yes. N -- _ -- 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] From: Unavailable sip:aster...@server.com; tag=as120a1079.
Hello Laszlo, That you for your response. Just to confirm callerid=whatever will only effect the private numbers? And will not have any effect on FROM headers with valid CIDs, as is intended? N. -- _ -- 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] From: Unavailable sip:aster...@server.com; tag=as120a1079.
Hello Everyone, Calls that are private name private number have the following TO header: From: Unavailable sip:aster...@server.com;tag=as120a1079. Don't tell anyone, but we are trying to put on a We're big enough to own the pricey softswitch look. Even though I would pick a OpenSIPS + Asterisk combo over a Metaswitch any day. Three words Service Licence Agreement. Anyhow long story short, is there any way to change the asterisk part only for the calls that are private? Everything else can stay the same.. Kind Regards, Nick. -- _ -- 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] From: Unavailable sip:aster...@server.com; tag=as120a1079.
Correction, and by TO, I mean FROM header :) -- _ -- 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] Dropped call on new CISCO router for no reason!
Hello Everyone, Just getting in a new cisco router, and would really like to get it up and running as soon as possible. Everything is configured from what we can see. This is a NAT setup. After 2 seconds on a successfully established call we reach retrans max, and asterisk disconnects the call. We have no idea why this is happening. SIP and RTP is flowing as expected. Your help is greatly appreciated, Nick. -- _ -- 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] Dropped call on new CISCO router for no reason!
Hello Eric, I knew this problem all so well however, never knew CISCO sip alg was enabled by default. The following settings got us up and going shortly after the email: no ip nat service sip udp port 5060 ip nat inside source static udp 192.168.2.5 5060 interface Dialer0 5060 access-list 130 permit udp any any range 8000 65535 route-map voip-rtp permit 1 match ip address 130 ip nat inside source static PRIVATE IP PUBLIC IP route-map voip-rtp Happy New Year to All, Nick from Toronto. -- _ -- 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] OT - Merry Christmas and a Happy and Prosperous 2014
God Bless and Merry Christmas to All! Nick. -- _ -- 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] Caller's phone keeps ringing after 200 OK
Hello Everyone, I have a strange issue where the caller's phone keeps ringing even after the 200 OK. I am using the latest version of Asterisk 1.8, and wanted to know if anyone could give me any pointers before posting the SIP debug messages. Kind Regards, Nick from Toronto. -- _ -- 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] file convert wildcard support
Hello Everyone, I was just wondering if the cli command file convert supports wildcard or entire directories? I am looking at a very long list right now and anxiously waiting a response :). Kind Regards, Nick from Toronto. -- _ -- 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] file convert wildcard support
DIR=sounds directory/*
cd $DIR
for f in $DIR; do
/usr/local/asterisk/sbin/asterisk -rx file convert $f ${f%.*}.g729
done
Thank you come again.
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[asterisk-users] Checking messages from outside the network
Hello Everyone,
I am using the following dialplan to allow users to check their
messages from PSTN world:
; Internal Routing
exten = _1XX,1,Dial(SIP/${EXTEN}, 10)
exten = _1XX,n,Wait(1)
exten = _1XX,n,Answer
exten = _1XX,n,Wait(1)
exten = _1XX,n,Voicemail(${EXTEN},us)
exten = _1XX,n,Hangup
The problem is that when the user presses `*#` to check his/her
messages, it adds
an additional message, even if there were no messages to begin with. I hope I am
explaining this correctly. Can the dialplan be improved so that there
is no additional
message added when the owner is trying to check their mail box.
Kind Regards,
Nick.
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Re: [asterisk-users] Checking messages from outside the network
Guys, Sorry, I pasted the wrong dialplan. What we are using is the following: exten = 6474770990,1,Answer exten = 6474770990,n,Wait(1) exten = 6474770990,n,Voicemail(1001,us) exten = 6474770990,n,VoicemailMain(1001) exten = 6474770990,n,Hangup So, we are using VoicemailMain, but wile trying to check messages (*#), it adds a message inadvertently. Can this be avoided? Nick. -- _ -- 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] Checking messages from outside the network
Thanks guys, it worked perfectly. No inadvertent message added when attempting to check messages from outside: exten = 1001,1,Answer exten = 1001,n,Wait(1) exten = 1001,n,Voicemail(1001,us) exten = a,1,VoicemailMain(1001) exten = 1001,n,Hangup Press * without pound, and voice mail main fires. Kind Regards, Nick from Toronto. -- _ -- 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] No remote address on RTP instance - On Ringing
Hello Everyone, I have a new problem where when placing the call, asterisk will automatically go into music on hold until the call is connected (ie, no ringing). It was kind of confusing because sometimes `SESSION PROGRESS` takes longer than others, during this time we are in MOH. The call does eventually connect and the MOH stops. When debugging I saw the following debug message: [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame This is a straight SIP channel. No DAHDI. Your Help is Greatly Appreciated. This is a new one for me :) Nick. -- _ -- 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] No remote address on RTP instance - On Ringing
There is two way audio, it's just during ringing that this happens. -- _ -- 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] No remote address on RTP instance - On Ringing
Hello Jg, Thank your for your response. No m option on dial. I think it's a RTP relay issue however, do not know how to diagnose the SDP payload. Any help would be appreciated. N. -- _ -- 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] No remote address on RTP instance - On Ringing
Yeah of course. Still digging into it :). Will post the solution if I find it. a2billing forum takes for ever to answer... -- _ -- 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] No remote address on RTP instance - On Ringing
I have no idea where the `m` is coming from. I even looked into the A2Billing script. Still digging -- _ -- 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] No remote address on RTP instance - On Ringing
Yes of course, I just did not want to overwhelm you guys with SIP trace. Before that though, I realized something: [Sep 10 12:03:30] WARNING[8178]: res_musiconhold.c:802 set_moh_exec: SetMusicOnHold application is deprecated and will be removed. Use Set(CHANNEL(musicclass)=...) instead -- AGI Script Executing Application: (DIAL) Options: (SIP/VTrunk/19042572451,60,HRrL(24:61000:3)m) There is that `m` option that jg was referring to. However, in a2billing, I have made sure there is no `m` option in the `Dial Command Params`: ,60,HRrL(%timeout%:61000:3). The extension for the entry does not include the option either: exten = 1000,1,Answer exten = 1000,n,Wait(1) exten = 1000,n,AGI(a2billing.php) exten = 1000,n,Wait(1) exten = 1000,n,Hangup Will run a test call with trace right now. Kind Regards, Nick. -- _ -- 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] No remote address on RTP instance - On Ringing
Hope this helps someone save a day of running around.
So my issue was with a2billing. The warning `No remote address on RTP
instance '0xb6d16a28' so dropping frame`
was not related to the music on hold coming on during ringing.
The Problem:
We have a script that loads rates into `a2billing.cc_ratecard` table. The
problem field was `musiconhold`.
Loading the field with `null`, causes this conditional statement in
`lib/asterisk/agi-bin/lib/Class.RateEngine.php` to fire:
if (strlen($musiconhold) 0 $musiconhold != selected) {
$dialparams .= m;
This added the m to the DIAL command.
The Solution:
Make sure cc_ratecard.musiconhold = '';
Thank you all for your help. I can rest now :).
Nick from Toronto.
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Re: [asterisk-users] No remote address on RTP instance - On Ringing
Oh scandalous Instead of playing the MOH, I would like to play the ringtone that is on the machine. Ummm, where is 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] No remote address on RTP instance - On Ringing
I ran a test call with trace can be found here: http://pastebin.com/f8MuxaFV I also wanted to mention that yes we have * setup with disallow=all and allow=g729 for testing, maybe permanently if we can successfully setup G729 pass through. That being said, the same problem is still there using allow=ulaw,alaw,g729. Thanks in Advance, Nick. -- _ -- 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] No remote address on RTP instance - On Ringing
Yeah!!! The Dial command setting: http://forum.asterisk2billing.org/viewtopic.php?f=2t=1704 I know this is not an a2billing mailing list, and I am sorry however, I do think that the No remote address on RTP instance may have something to do with it. Maybe the ringtone from downstream is not reaching asterisk, and thus a2billing is appending the `m` to the dial command? I'm sorry if this sounds crazy... :) N. -- _ -- 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] G729 Passthrough How To
You ok sir? Are you going to make it? N. -- _ -- 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
