Re: [asterisk-users] Forking AGI or GoSub
On 4/19/2019 1:49 PM, Dovid Bender wrote: > Mark, > > I am using PHP agi and when forking the call does not continue util > the forked process is done. Am I doing it wrong? > > > On Wed, Apr 10, 2019 at 4:27 PM Mark Wiater <mailto:mark.wia...@greybeam.com>> wrote: > > On 4/10/2019 3:54 PM, Dovid Bender wrote: >> I have an AGI that can sometimes take time complete. I don't want >> the dialplan to be held up by the agi. Is there any way to call >> it and have Asterisk continue with the dialplan? >> > > Is there a reason you can't fork in the AGI and just return to the > dialplan in the parent? > Dovid, I'm not much of a PHP person, but in perl, i check the process id that's returned from fork() and exit if it's 1 (parent) and keep processing if it's the child (greater than 1). I think php uses pcntl_fork(). Is that how you're doing it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forking AGI or GoSub
On 4/10/2019 3:54 PM, Dovid Bender wrote: > I have an AGI that can sometimes take time complete. I don't want the > dialplan to be held up by the agi. Is there any way to call it and have > Asterisk continue with the dialplan? > Is there a reason you can't fork in the AGI and just return to the dialplan in the parent? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd one-way audio problem (Mike Diehl)
On 3/25/2019 4:45 PM, Mike Diehl wrote: > > > So, I don't think it's their network. I've taken pcaps of both legs of > > > example calls. On the provider-side, I see 2-way audio. On the > > > client-side, I only hear one side. > Mike, In those pcaps, are you seeing the exact same RTP traffic between provider side and client side? And was client side captured close to the phone, past the firewall if there is one? Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx / Asterisk PJsip multipe devices
These two phones are not using the same extension, are they? On 2/6/2019 8:49 AM, basti wrote: > both phones are registered. and the hardware phone can also make calls. > but an incoming call is not displayed and also not hearing. > > Call Waiting is also disabled. > > On 06.02.19 14:07, Cyril Alberts wrote: >> Hi, >> look at your registrations, is the hardware phone registered? >> if yes, which phone vendor do you want to connect? can you make >> outgoing calls with hardwarephone? >> >> BR Cyril >> >> Am Mittwoch, den 06.02.2019, 13:00 +0100 schrieb basti: >>> Hello, >>> >>> I have some user that had have a hardwarephone and an softphone. I >>> use >>> pjsip driver and set "Max Contacts = 2" to have register both at the >>> same time. >>> >>> But Only the softphone is ring. the hardware phone is mute. >>> >>> How can i fix this? >>> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Timestamp rewind
On 8/30/2017 5:03 AM, Steve Davies wrote: > Mark, > > You have cropped the image you inserted above and removed a very > important part of the line you highlighted. I think is says ",Mark" > after the time value - You can even see the un-cropped comma in your > picture. Thanks Steve, I did omit the Mark indication in the screenshot. > > RTP timestamps can be reset mid-stream if needed - It is part of the > spec, and most commonly happens when initially (eg Asterisk) generated > audio is replaced with audio from an external source once the call is > bridged. The early timestamp comes from Asterisk, and the subsequent > timestamp is retained from the new source of the RTP. Thanks! That helps. I had read the portion of rfc3550 that said The sampling instant MUST be derived from a clock that increments monotonically and linearly in time to allow synchronization and jitter calculations So that led me to believe that the timestamps should increment at predictable intervals. Wireshark flagged it in the RTP stream analysis as being an Incorrect Timestamp too. > No packets should be dropped though in my experience some jitter > buffers can handle it poorly. > > Hope that helps, > Steve > Thanks for your response. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP Timestamp rewind
Hi folks. I have a couple of questions regarding RTP. The background of my inquiry is that I have packet captures of SIP and RTP traffic on an Asterisk and Broadworks SIP trunk and the RTP many times has a time stamp that rewinds by 480 using g.711u. The Sequence number continues to increment appropriately, but the timestamp just rewinds. It doesn't happen on every call, but it's frequent enough to make me want to understand it better. My questions are: Is there ever a circumstance where it would be normal or logical to see the RTP timestamp go backwards during the RTP stream? Consistently by 480, 3 voice frames? Will Asterisk just drop the packets that compromise the rewind? Thanks Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Integration of Google Speech API V2
I've had Lefteris' code running for a few years without a problem. I don't have a service key but I have entered my API key in the script in the 'User defined parameters' section. You did that, right? What do the other user defined parameters in your script look like? On 7/19/2017 4:37 AM, Rahul MathuR wrote: > Hi, > > I'm trying to integrate Google cloud speech recognition v2 in it. I > can get the audio recorded, have created Service key and API key but > whenever I try to access it, I just get 403 access denied. I am at my > wits end here. > > Has anybody tried it ? were you successful ? Could you please guide me > how to do it ? > I'll be grateful to you if this works ! > > > > -- > Warm Regds. > MathuRahul > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Copying received and sent RTP packets due legal obligations
On 7/12/2017 5:30 PM, Holger Freyther wrote: > I have to copy/mirror/forward the RTP streams for some selected call > to an external address/port I'd think that what you want to do might be best done outside of Asterisk. If you're working with SIP, I'd suggest packet capture tools. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Writing CDR's to two database servers
On 6/20/2017 8:42 AM, Tech Support wrote: > I appreciate all the feedback, and replication seems to be a logical > solution, but I was initially thinking about how to implement a solution > within Asterisk to write the CDR's to two databases. Is that possible? Now > I'm just curious. Sorry, maybe it's been mentioned. An AGI at call termination to write to the other database? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Want to capture all SIP messages
On 5/31/2017 3:36 PM, Steve Edwards wrote: I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? voipmonitor is what you want. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatically dial a number, then an extension
On 5/25/2017 11:11 AM, Tech Support wrote: I need to be able to tell whether or not the far end extension picked up might a waitForSilence come in useful here? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.
On 4/18/2017 7:40 PM, Ernie Dunbar wrote: Server network: 192.168.0.0/24 OpenVPN network: 10.8.0.0/24 Asus network: 192.168.1.0/24 The Asterisk SIP registration appears to be responding properly to this - this is what I see when I do a 'sip show peer' for an Aastra phone that's connecting through the VPN (Asterisk output is truncated): ToHost : Addr->IP : 10.8.0.6:5060 If the Asus network is 192.168.1.0/24, and the phone is registering as 10.0.8.6, it looks like NAT is taking place. Would your asterisk server know how to route traffic to 192.168.1.0/24? I've always used site-to-site OpenVPN tunnels where the vpn's terminate on the gateway for both the phones and the asterisk server. I've always had rock solid connections between phones and Asterisk. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to send SIP_NOTIFY messages with variable content ?
On 1/18/2017 9:58 AM, Tech Support wrote: For reconfiguring SIP phones? Can you give an example or short explanation? One can send a SIP notify with a check-config to the phone and have the phone re-download it's configuration files from a provisioning server. In the CLI, you can do a SIP NOTIFY with one of cisco-check-cfg, grandstream-check-cfg, polycom-check-cfg, sipura-check-cfg, snom-check-cfg and the extension. I've done it with Yealink phones too, don't have the proper syntax in front of me though. Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SoundStation IP 6000 does not register
Yves, Didn't you say that AsteriskServer: 192.168.1.211 SIP-user: 165 ? On 12/21/2016 4:24 AM, Yves wrote: . It is sure for 100% that there is no firewall or something else mangeling in between... another Hardphone works as expected using the same Netzworkcable on the same Networkplug with UDP on Port 5060... This other hardphone, what IP does it have? 50.848|cfg |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask 255.255.255.0 The line above suggests to me that your phone and your asterisk server are on a different network, there has to be something that routes between those two networks. Often what routes, can firewall. 000122.941|sip |4|03|Registration failed User: 165, Error Code:480 Temporarily not available Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SoundStation IP 6000 does not register
On 12/19/2016 10:26 AM, Yves wrote: There are no SIP Packets arriving at my asterisk at all... and it has nothing to do with a firewall or similar... I can ping the phone from the asterisk, If both of these items are true, then I'd look at the phone configurations. Does the provisioning file contain an address for the phone to contact? Mine has voIpProt.server.1.address, but I think you can also use a reg.x.address in the provisioning files too. Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk inside network. What phone works well?
I think you had asked what phone works well with VPN's. I've had very good experiences with Yealink using OpenVPN, never an issue. I think I've heard that Snom does OpenVPN as well. Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple phones when one is unregistered
On 9/1/2016 6:55 AM, D'Arcy J.M. Cain wrote: So does the Dial command go directly to the registered device or does it use the extension? Yes, that's why you provide the technology part (SIP/, IAX/, DAHDI/) I was assuming that it was going to the extension's voice mail if it wasn't there but that's in the extension dialplan and I suspect that the extension is irrelevant and only the SIP registration matters. That would be a good thing since many extensions also ring the user's cell phone and that would be annoying if they were at home when someone came to the office door. If the target device has a forward configured, I believe Asterisk will dutifully ring the forward number like it was told, but could be wrong. Otherwise, it'll only go to voicemail, in asterisk, if you tell it to in the dialplan. Could it be going to voicemail on a forward somewhere? Perhaps my problem was that one of the users was removed from sip.conf but their phone was still in the above plan. In my experience, that too should not be a problem. I've got the same dial command structure, ringing multiple SIP/extensions in one dial command. I have some instances where the SIP entry in sip.conf no longer exists, and some instances where the device is not registered, and asterisk just displays a warning and continues on. For example, user2 leaves the company, is removed from sip.conf but we forgot to remove him from the door buzzer extension. That might give me the behaviour that I was seeing. I don't think so. At least I don't see that. Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple phones when one is unregistered
On 8/31/2016 9:57 PM, D'Arcy J.M. Cain wrote:
exten => 55,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN
USE"]?Set(toRing=${toRing}/user1)
Failed. I checked the online docs and the syntax seems to be correct
but I get this:
[Aug 31 21:52:00] WARNING[-1][C-0001fed5] pbx.c: No application
'ExecIf' for extension (unauthenticated, 55, 3)
Set is a function, not an application. ExecIF executes an application.
I'm a bit confused by this whole topic. The dialplan snippet in the
original email
exten => 55,1,Verbose(Door buzzer calling)
same => n,Dial(SIP/user1/user2/user3)
should have rung the phones forever as long as one phone was active and
not forwarding or DNDing.
Mark
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Re: [asterisk-users] Windstream SIP Trunk settings
In my case, username is the BTN. I also set the fromdomain to be the sbc that I'm registering with. Externip might help also? [paetec] host=10.250.0.5 username=btn fromdomain=10.250.0.5 dtmfmode=rfc2833 externip=10.255.0.2 I've used these settings on both registering and non-registering trunks, connecting to both the Broadworks and Plexus platforms in Windstream. Though all of my asterisk versions have been 1.8.x Mark On 2/22/2016 8:20 AM, James Cass wrote: > Does anyone on this list use Windstream as a SIP trunk provider? > > If so, would you mind sharing your peer settings? > > I'm using asterisk 13.7.2 and can't seem to get the inbound working > correctly (using registration). Outbound is fine, but they are seeing > an authentication error on their end. > > Here are my inbound peer settings: > > username= > secret= > host= > type=peer > fromuser= > context=from-trunk > dtmfmode=auto > canreinvite=no > qualify=yes > insecure=port,invite > > register string: :@:5060 > > Thanks in advance, > > -- _ -- 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] Calendar integration : Could not authenticate to server: rejected Basic challenge
On 10/27/2015 8:56 AM, Jonas Kellens wrote: > > I have changed this setting at Google but it brings me no success. > Jonas, I've been using google calendar and Asterisk 1.8 for a couple of years now without issue. I have a note in my configuration that says that I'm using the Private ICAL URL from gmail and that it's the only one that worked for me. Is that the URL that you're using? Did you change your type to ical in calendar.conf? Mark -- _ -- 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] setting outbound caller ID
On 6/18/2015 1:27 PM, Greg Woods wrote:
My provider claims that I am somehow sending an old number that
doesn't appear anywhere
snip
(I just moved from a POTS provider Century Link to a VOIP provider).
Set(CALLERID(number)=${var}) works fine for me.
Perhaps some debugging on the channel could help? Set sip debug if it's
a sip trunk?
You'll at least get to see the callerid that Asterisk is putting on the
trunk. That might even help your new VOIP provider do some digging if
could provide the debugging output.
Mark
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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?
On 10/28/2013 3:59 PM, Ron Wheeler said: I am reaching the same level of frustration. I have tried to find the source of the problems. We have IAX2 to our VoIP provider and SIP phones attached to the Asterisk - No analogue. I don't have any problems with IAX, but I hear some do. We have a very lightly loaded 60 Mbs cable link to the Internet that tests pretty close to that most of the time. Bandwidth is less important than the overall quality of the internet link, latency and jitter. Either way, there is no QoS on the internet, all bets are off. The codec can matter too. What are you using? I have not found any good tools to track down the causes of poor voice quality. In my case, I have good incoming quality and terrible quality going out. Oh, is your cable connection assymetric? Upload smaller than download? If so, that correlates to terrible audio, right? That is, I can hear people perfectly well but they complain that my voice drops out and is garbled regardless of who places the call. As a result, I use Skype for all of my calls and if someone calls me, I call them back on Skype if they have any problems. I don't understand why Skype works so well and Asterisk works so poorly on the same environment. Googling Asterisk poor audio quality return several hundred thousand references I'd not shoot asterisk yet. I'd focus on the internet connection and it's components (cable modem) first. I use asterisk all over the place. Mostly connected to PRI's and Carrier provided SIP trunks, with internet SIP trunks as backup. I get complaints on the Internet based SIP trunks sometimes, never on other other two. I'd ask most of these questions of the OP too. Overall telephony design matters. -- _ -- 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] Function not Registered??
On 5/25/2012 3:18 AM, Lee, John (Sydney) said: -- Executing [*1223*1**1900@incoming:78] Set(SIP/1900-08ee1da8, DEVSTATE(Custom:cfalw1900)=INUSE) in new stack I use 'Set(DEVICE_STATE(Custom:var)=BUSY)' in my 1.4 dialplans to set device state. mark -- _ -- 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] Mitel integration
the mitel 3300 sends SMDR on TCP 1752. It spews software and hardware logs in the same manner, different ports. On 1/27/2010 11:00 AM, Steve Howes said: On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote: Sounds good to me, but without the spec I'm stuck in a catch 22! tcpdump? (assuming IP). Bet its fairly simple plain text or something. Steve attachment: Mark_Wiater.vcf-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Reply to Our Critical Packet SIP Calls Dropped in Voicemail
Lincoln King-Cliby wrote: -Original Message- Then starting at packet 3217 there are a series 6 of ICMP Destination unreachable (Port Unreachable) messages from the Asterisk server to the phone, with an RTP packet from the Phone to the Asterisk server before each Destination unreachable message. Wouldn't this suggest that either Asterisk couldn't open the port, or opened it and then closed it? Or I suppose that perhaps the phone and asterisk didn't negotiate the port properly? Does your packet capture show that the phone is consistently using the correct port to communicate with the * server? There's no change or anything, right? You don't happen to have a corresponding sip debug to this wireshark capture do you? You might be able to correlate info from the two. In your original post, I thought I read that you could reproduce this issue by increasing load on the asterisk server. What does the caller experience in the first 20 seconds when a call to voicemail is going to fail? Just ringing? Any chance there's anything in Asterisk's or the OSes logs about some failure of the network stack? What OS is this? Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] snap a number now digium?
isn't snapanumber turning into ADA? http://dl1.digium.com/ADA/ADA_MIS.pdf There's a forum too. Looks an awful lot the same. Can't get it to work with my Thunderbird contacts the way snapanumber does though. Maybe it's a work in progress? Mark Dean Collins wrote: Wow they must have bought the company. I use the software and love it - shame to see it disappear. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: Wednesday, 21 January 2009 1:15 PM To: Asterisk Users Mailing List Discussion Subject: [asterisk-users] snap a number now digium? Where's it gone? Going to http://www.snapanumber.com/ goes directly to the digium site with no indication of where it is ... Has it gone forever? Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Returning to Voicemail after returning call
Hello all, I've got dialout= and callback= set in my voicemail.conf so that I can have users return calls to folks who have left messages. They really like this feature. But when the callback is over, a normal hangup occurs instead of the caller being put back into voicemail at the next message. Is it possible that the users be returned into the voicemail system where they left off? thanks Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones
Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the Mitel. Or so it seems. The receiving Mitel phone thinks the call is in session though. Oh... this does not happen all of the time, maybe 50%. Asterisk is reporting errors like: [Jul 19 10:46:41] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '72.16.1.20;tag=as7b9f4bfb' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' or [Jul 19 10:45:03] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '172.16.1.20;tag=as4a1b11c8' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:45:03] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: can't resolve in DNS) : '172.16.1.20' [Jul 19 10:45:04] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '172.16.1.20' [Jul 19 10:45:04] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '172.16.1.20' or [Jul 19 10:52:18] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: 'nt-Length:0' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:52:18] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'nt-Length' [Jul 19 10:52:18] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host 'nt-Length' [Jul 19 10:52:18] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host 'nt-Length' -- SIP/517-09215fb0 answered SIP/512-09258c78 -- Native bridging SIP/512-09258c78 and SIP/517-09215fb0 [Jul 19 10:52:18] WARNING[22054]: chan_sip.c:5839 set_destination: Can't find address for host 'nt-Length' -- Got SIP response 416 Unsupported URI Scheme back from 172.16.1.157 [Jul 19 10:52:18] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host 'nt-Length' So it seems that the Mitel phone is sending a bad contact field in SIP. I've confirmed via tcpdump that this is what's in the SIP packet on the wire. I wanted to try a different version of SIP on the Mitel but that doesn't seem to be an option, it's not available for download and the local Mitel vendor can't seem to get his hands on anything newer than 6.0.0.something, though there is supposedly 7.1.x available. These phones are running 06.00.00.19. The Asterisk server has a pretty standard sip.conf, bindaddr=0.0.0.0 pedantic=no; bindport=5060 srvlookup=no tos_video=af41 notifyringing=yes notifyhold=yes allowsubscribe=yes limitonpeer=yes localnet=172.16.1.0/255.255.255.0 Polycom phones on this same asterisk server do not display this behavior. I'm wondering if there is a workaround for this apparent Mitel issue in Asterisk's configuration. Anyone using this combination with success? Thanks in advance for any thoughts Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not a valid SIP contact - Asterisk 1.4.21.1 Mitel SIP phones
Matt Watson wrote: On July 19, 2008 11:22:08 am Mark Wiater wrote: Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the Mitel. Or so it seems. The receiving Mitel phone thinks the call is in session though. Asterisk is reporting errors like: [Jul 19 10:46:41] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '72.16.1.20;tag=as7b9f4bfb' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '72.16.1.20' Might want to post a sip debug of one of the sessions from the Mitel phone. Thanks Matt I was also able to test this with Mitel's firmware version 7.0.0.8 with the same results. Mitel phone still acts like it's on a call, Asterisk does not nor does the originating phone. PBX*CLI sip set debug peer 517 SIP Debugging Enabled for IP: 172.16.1.174:5060 Audio is at 172.16.1.20 port 15594 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.16.1.174:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 172.16.1.20:5060;branch=z9hG4bK38581b5a;rport From: 512 sip:[EMAIL PROTECTED];tag=as7ec9e8af To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 19 Jul 2008 17:20:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 236 v=0 o=root 2247 2247 IN IP4 172.16.1.20 s=session c=IN IP4 172.16.1.20 t=0 0 m=audio 15594 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 100 Trying Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED] Content-Length:0 - --- (8 headers 0 lines) --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 180 Ringing Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED] Allow-Events:talk,hold,conference Content-Length:0 - --- (9 headers 0 lines) --- PBX*CLI --- SIP read from 172.16.1.174:5060 --- SIP/2.0 200 OK Via:SIP/2.0/UDP 172.16.1.20:5060;rport=5060;received=172.16.1.20;branch=z9hG4bK38581b5a From:512 sip:[EMAIL PROTECTED];tag=as7ec9e8af To:sip:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User-Agent:Mitel-5212-SIP-Phone 07.00.00.08 08000F32A6C1 Call-ID:[EMAIL PROTECTED] Contact:p:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 CSeq:102 INVITE User sip:[EMAIL PROTECTED] Allow-Events:talk,hold,conference Allow:INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY,PRACK,UPDATE Supported:timer,100rel,replaces Content-Type:application/sdp Content-Length:182 v=0 o=517 1216473942 1216473941 IN IP4 172.16.1.174 s=SIP Call c=IN IP4 172.16.1.174 t=0 0 m=audio 20012 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 - --- (15 headers 8 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 172.16.1.174:20012 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.1.174:20012 [Jul 19 13:20:56] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: 'p:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 13:20:56] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '172.16.1.174' list_route: hop: p:[EMAIL PROTECTED];tag=4881ea36-2ca-6747d965 set_destination: Parsing p:[EMAIL
Re: [asterisk-users] Shared line appearance phones?
Russell Bryant wrote: Ron McCarthy wrote: Asterisk 1.4 im guessing? I did not know the Snom's worked with that, Ill have to check it out then! The way it is implemented in Asterisk is a bit interesting. It uses the existing device state support (hints, BLF) to manage the buttons for shared lines. Asterisk changes the state of these virtual shared lines to different states, and the light on the phone reflects the state (in use, ringing, on hold). I fought with this in 1.4.5 with polycom phones. I was hoping to share a DID from a PRI on several Polycom IP430's. Might you be willing to share some specific configurations for such a situation? thanks mark ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] linksys SPA-941
I asked the [EMAIL PROTECTED] for the documents and the tools that are referenced in the admin guides and was told that I had to become a registered user in the support section of the ww.sipura.com website. They wanted name, title, phone # and type of support I provide for the devices. I think I actually became registered via email with [EMAIL PROTECTED] Got an email the next day with user information for their support site. mark Edwin Lam wrote: does anyone get a hold of the SPA-941 Provisioning Guide? i tried call Sipura's tech support, seems like none of them heard of the term remote provisioning. they kept refering me to their web site which i've check thoroughly, and could not find any documentations on the SPA-941. finally they gave me a phone number to call, which appears to be a fax machine. that's when i gave up on those idiots. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
