Re: [asterisk-users] Asterisk stopping for no reason
Hi, please always add asterisk version to your query. I managed to run internet radio (that streams MP3) within asterisk. Minor change is nescesarry to make it work with random MP3s. My Dialplan: exten = _X.,n,Answer() exten = _X.,n,MP3Player(http://stream.m-1.fm/m1/mp3) $ cat /usr/bin/mpg123 #!/bin/bash /usr/bin/wget -q -O - $1 | /usr/bin/madplay -Q -z -o raw:- --mono -R 8000 -a -6 - You should change the WGET part to something that better suits your needs. Tested on asterisk 1.4.27 On Thu, Apr 29, 2010 at 10:59 PM, Alexandre Vézina avez...@vencomm.ca wrote: Hi, Few days ago, my asterisk began to stop unexpectedly What I did: Added a mp3 to the musiconhold directory Adjusted the permissions (chown asterisk:asterisk + chmod 755) Reconfigured the musiconhold.conf to the deprecated format (found the example on the internet) [classes] default = quietmp3:/etc/asterisk/moh,r Restarted the service I thought the new mp3 was corrupted so I removed it from the server. The problem perssisted so yesterday I changed the deprecated configuration to: [default] mode=quietmp3 directory=/etc/asterisk/moh random=yes My original configuration was: [default] mode=files directory=/etc/asterisk/moh I have no logs telling me thate quietmp3 failed and I cannot find any way to see if the musiconhold was enabled when asterisk dropped. Here are my questions (finally): Do you know if quietmp3 may kill the server? Is there a way to set random in files mode? I am using Asterisk 1.4.17~dfsg-2ubuntu1.1 on an Ubuntu 8.04.4 server. Thank you very much --- Alexandre Vézina -- _ -- 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] ATA shootout: PAP2T versus Grandstream Handytone 286
Hi, I had to choose between an 8 port FXS device from Cisco/Linksys (the sipura 3000) and a similar device from Grandstream. A look on the Grandstream's forums had me scratching my had, so much people with problems, frequently needed restarts, etc. The next thing, the Cisco/Linksys seems to be manufactured (at least this device) with durability in mind, it includes a Fan and a sturdy aluminium case, wheres the Grandstream was plastic and as far as I recall had no cooling. I went with the Cisco because I really needed this thing stable (it's on a Golfcourse and having there a problem means long car-drives) so up until know (nearly a month in production) I have no Problem at all. Way cheaper and easier to deploy than an internal FXS card ... So, do yourself a favor and test the devices sometimes 15 bucks per device might seem at first a bargain, but if you have more problems in the long-run sometimes it turns into exactly the opposite. best regards Ray - RunSolutions Open Source It Consulting - Email: r...@runsolutions.com Parc Bit - Centro Empresarial Son Espanyol Edificio Estel - Local 3D 07121 - Palma de Mallorca Baleares - Mensaje original - De: David Backeberg dbackeb...@gmail.com Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Enviados: Jueves, 29 de Abril 2010 21:36:36 Asunto: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286 I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an alternative to the PAP2T, and I'm seeing prices hovering between $25 and $30. I'm considering getting one of these Grandstream ATAs onsite to play with before I make my final decision. What do people think about both products? Bonus points for if people have bulk deployed these, either with TFTP and configs pushed from a server, or some other good idea. It seems that the PAP2T does support TFTP and an XML-based config for deployments... -- _ -- 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 and Patton
Hi, we have and Asterisk server connected to a Patton Smartnode 4638 with 4 BRI. We configured 4 SIP account on Patton (1001, 1002, 1003, 1004). The system is fully functional, but we have a problem to recognize incoming calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton) or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming from SIP/1004. I have contacted Patton support, I have send configuration and debug and they told me that there is a problem of Asterisk configuration. In the sip debug on Asterisk I have seen (SIP/1001 incoming call) ... Sending to 192.168.2.122 : 5060 (no NAT) Using INVITE request as basis request - 89c9689349c54649aae566e9192c5...@192.168.2.122 Found peer '1004' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 ... (192.168.2.122 is the ip address of Smartnode.) In the Patton configuration ... gateway sip ASTERISK bind interface LAN router service default defaultserver manual 192.168.2.121 5060 loose-router registration manual 192.168.2.121 user 1001 authenticate password 36ocYTYpKxk= encrypted register display-name 1001 gateway sip ASTERISK no shutdown ... (192.168.2.121 is the ip address of Asterisk server) The call is coming from SIP/1001, but the INVITE request founds peer 1004. The problem come when I try to use FOP: I am not able to correctly connect button to trunk. Someone can help me? Thanks in advance. Eco -- _ -- 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 and Patton
Il giorno 30/apr/10, alle ore 10:01, A.Santoro ha scritto: Hi, we have and Asterisk server connected to a Patton Smartnode 4638 with 4 BRI. [...] Hi Eco, I think the problem is in your sip.conf. Have you tried setting insecure=port,invite in the sip.conf for each sip account? Bye, Carlo -- _ -- 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 and Patton
2010/4/30 A.Santoro n...@ecoricerche.it Hi, we have and Asterisk server connected to a Patton Smartnode 4638 with 4 BRI. [...] Hi Eco, I think the problem is in your sip.conf. Have you tried setting insecure=port,invite in the sip.conf for each sip account? Bye, Carlo -- _ -- 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] ATA shootout: PAP2T versus Grandstream Handytone 286
--- On Fri, 4/30/10, Raimund Sacherer r...@runsolutions.com wrote: Hi, I had to choose between an 8 port FXS device from Cisco/Linksys (the sipura 3000) and a similar device from Grandstream. A look on the Grandstream's forums had me scratching my had, so much people with problems, frequently needed restarts, etc. The next thing, the Cisco/Linksys seems to be manufactured (at least this device) with durability in mind, it includes a Fan and a sturdy aluminium case, wheres the Grandstream was plastic and as far as I recall had no cooling. I have quite a few Grandstream GXW4008 devices and I must say that early firmware versions were a disaster. However, it's been at least a year now that I'm running these devices with no major problem with their latest firmware. I'm not biased and must say that they're stable now. I also have a Linksys SPA8000 (8-port ATA equivalent) with internal fan, etc., but despite its stability I've had a few non-critical issues with transfers and early dials. I must say however that support is a tad better in Grandstream than Linksys. As far as having an internal fan for cooling, I don't know if that's actually better... In general, these devices shouldn't need to rely on mechanical cooling which tends to fail in time (sure, you can open the case and replace it but that's extra maintenance). Vieri -- _ -- 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] ATA shootout: PAP2T versus Grandstream Handytone 286
On 30 Apr 2010, at 09:41, Vieri wrote: As far as having an internal fan for cooling, I don't know if that's actually better... In general, these devices shouldn't need to rely on mechanical cooling which tends to fail in time (sure, you can open the case and replace it but that's extra maintenance). The fan in the 8000 and the 8800 is horribly loud. Taking the screws out and mounting it on sticky pads helps.. S -- _ -- 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] Call-Waiting, implementation ideas
Hi all, How can I implement a full-featured Call-Waiting behavior on the Asterisk level (e.g. I don't want to relay on end-equipment capabilities)? I found it very strange that such a basic feature is not built-in in Asterisk (and I've googled a lot in search for this). Here is what I need: SomeuserX is calling MyUserA. They are on conversation (assumption: voice is via the Asterisk) SomeuserY is calling MyUserA. SomeuserY gets a special ringing tone. Meaning - Asterisk opens voice channel towards SomeuserY (progress with SDP) and plays SpecialRingBack.wav/gsm etc. MyUserA Gets voice notification (e.g. beep-beep) during his call to SomeuserX. Meaning - Asterisk barge-in the rtp stream and play the file beepbeep.wav/gsm on the MyUserA channel. This is done periodically for as long as SomeuserY is waiting to be answered (i.e. doesn't hang-up). Asterisk is monitoring the state of the call SomeuserX - MyUserA. If MyUserA will signal (e.g. hook-flash or some digit sequence) that he wants to answer the 2nd call then Asterisk will put on hold SomeuserX and bridge SomeuserY to MyUserA with the option for MyUserA to toggle between the two channels. If the conversation SomeuserX with MyUserA is terminated Asterisk will INVITE MyUserA and when picked up will bridge SomeuserY with MyUserA. I hope there is a solution for that… I tried using DEVICE_STATE for this purpose however I keep getting status NOT_INUSE even if the extension IS in use (I'll open a different thread on this issue if needed). Thanks in advance for any ideas provided, Harel -- _ -- 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] ATA shootout: PAP2T versus Grandstream Handytone 286
In my previous company we bought about 30 Grandstream GXP2000 phones. The build and design quality of those phones were terrible (not to mention firmware bugs). Speakerphone and headset ports were unusable. The external powersupply would only last a year or two before it failed. The screen was so poor it became very dark after the backlight had been on a few months. The internal PSU for taking power via POE used cheap components and within 2-5 years every phone had failed. Luckily I worked out how to fix them. Newer models may be much better but I would not buy another one again. David Backeberg wrote: I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an alternative to the PAP2T, and I'm seeing prices hovering between $25 and $30. I'm considering getting one of these Grandstream ATAs onsite to play with before I make my final decision. What do people think about both products? Bonus points for if people have bulk deployed these, either with TFTP and configs pushed from a server, or some other good idea. It seems that the PAP2T does support TFTP and an XML-based config for deployments... -- _ -- 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] Problems with t38modem and bitrate sent to t38-termination service
Hi all the people in the list!
I'm new on this list, this is my first post.
I configured asterisk 1.6 with freepbx 2.7 and dahdi to send faxes with
t38modem conected to hylafax as a sip extension of asterisk.
Everything is supposed to be configured fine, the faxes start sending, but
at the middle of the transaction, it fails. The T.38 termination provider
told me that they were receiving a=T38MaxBitRate:2400 on their asterisk, and
told me that this issue could be the problem.
I've checked all the configuration and can't see any kind of configuration
for the bitrate of the t38modem.
I don't know if this is an Asterisk issue or a t38modem issue.
My configurations are:
/etc/asterisk/udptl.conf:
[general]
udptlstart=4000
udptlend=4999
udptlchecksums=no
T38FaxUdpEC = t38UDPRedundancy
T38FaxMaxDatagram = 400
udptlfecentries = 3
udptlfecspan = 3
use_even_ports=no
/etc/asterisk/sip_custom.conf:
[t38modem-options](!)
type = friend
host = 127.0.0.1
context = fax-out
canreinvite = no
disallow = all
allow = ulaw
t38pt_udptl = yes
t38pt_rtp=no
t38pt_tcp=no
dtmfmode = rfc2833
nat = no
qualify = yes
[T38modem0](t38modem-options)
port = 6060
[T38modem1](t38modem-options)
port = 6061
[T38modem2](t38modem-options)
port = 6062
[T38modem3](t38modem-options)
port = 6063
[T38modem4](t38modem-options)
port = 6064
/etc/asterisk/extensions_custom.conf:
[fax-out]
exten = _XXX,1,Dial(SIP/991321604${ext...@t38faxing,,R)
exten = _X.,n,Hangup()
/etc/asterisk/sip_additional.conf:
[t38faxing]
disallow=all
allow=ulaw
canreinvite=yes
host=xx.xx.xx.xx
outboundproxy=xx.xx.xx.xx
fromdomain=9913216.paygvoip.com
insecure=port,invite
type=peer
dtmfmode=rfc2833
username=userid
password=mypassword
fromuser=userid
nat=yes
context=from-trunk-sip-t38faxing
I'm getting crazy with this, I don't know where the problem could be. Any
kind of help would be appreciated.
Thanks a lot for your time and Implication on this list.
Regards,
Miguel Amez
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[asterisk-users] IAX trunks and audio codecs
Hi, I have IAX trunks between Asterisk servers. They receive calls on ISDN cards and Dial() through the IAX trunks to the primary Asterisk server where all the SIP phone extensions are registered. The IAX trunk settings are something like this (all servers have this identical except for the host field): [inbound] deny=all allow=alaw allow=gsm type=friend host=192.168.250.111 secret=inboundpass auth=plaintext requirecalltoken=no qualify=yes context=from-inbound username=inbound trunk=yes I'm trying to force the use of alaw because some of the local SIP extensions use this codec (a minor percentage use gsm) and none use ulaw. So I suppose that if the first Asterisk server that receives the call and sends it out to the main server via IAX encodes in alaw then the main server won't have to transcode if the destination is also alaw (most SIP phones). This should save some CPU processing in the main Asterisk server, right? So my trouble is with this message on the main Asterisk server when it receives a call from a secondary server via IAX: Apr 30 12:19:59] NOTICE[14517] channel.c: Dropping incompatible voice frame on IAX2/inbound-2255 of format alaw since our native format has changed to 0x4 (ulaw) Why is it changing to ulaw if I'm explicitly allowing only alaw and gsm and denying the rest? Thanks, Vieri -- _ -- 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] HDLC Receiver overrun on Wildcard TE410P
Hello I've got small PBX (30 simultaneous connections) based on asterisk (1.6.2.6), which uses Stargate 2N ISDN to GSM gate. It runs ok for day or two, but then I get: dahdi: HDLC Receiver overrun on channel TE4/0/1/16 (master=TE4/0/1/16) in my kernel logs, in asterisk i get: pri show spans PRI span 1/0: Provisioned, Down, Active PRI span 3/0: Provisioned, In Alarm, Down, Active (span 3 is not connected to gateway for now) and I can't make any calls. My dahdi-channels.conf: ; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_net channel = 1-15,17-31 context = default group = 63 /etc/dahdi/system.conf: # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,0,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 /proc/interrupts: CPU0 CPU1 CPU2 CPU3 0:462313432 0 IO-APIC-edge timer 1: 3 5 5 3 IO-APIC-edge i8042 8: 32 31 32 34 IO-APIC-edge rtc0 9: 0 0 0 0 IO-APIC-fasteoi acpi 12: 28 27 29 30 IO-APIC-edge i8042 14: 3 3 1 0 IO-APIC-edge ata_piix 15: 0 0 0 0 IO-APIC-edge ata_piix 16: 0 0 0 0 IO-APIC-fasteoi uhci_hcd:usb2, uhci_hcd:usb5 18: 0 0 0 0 IO-APIC-fasteoi uhci_hcd:usb4 19: 0 0 0 0 IO-APIC-fasteoi uhci_hcd:usb3 23:351350200188 IO-APIC-fasteoi ehci_hcd:usb1 25: 0 0 0 1 IO-APIC-fasteoi 26: 2151 2083 104408008 1985 IO-APIC-fasteoi eth1 51:10899621089895 2940 2910 IO-APIC-fasteoi cciss0 78: 485362 485513 485473 320148527 IO-APIC-fasteoi wct4xxp NMI: 0 0 0 0 Non-maskable interrupts LOC: 30245307 31461472 20982472 23492748 Local timer interrupts SPU: 0 0 0 0 Spurious interrupts CNT: 0 0 0 0 Performance counter interrupts PND: 0 0 0 0 Performance pending work RES: 436674 44030221952681451020 Rescheduling interrupts CAL:169265203251 Function call interrupts TLB: 43920 44257 50177 52884 TLB shootdowns TRM: 0 0 0 0 Thermal event interrupts THR: 0 0 0 0 Threshold APIC interrupts MCE: 0 0 0 0 Machine check exceptions MCP: 1073 1073 1073 1073 Machine check polls ERR: 3 MIS: 0 I manually set irq affinity - eth1 to CPU2, digium card to CPU3, rest of common interrupts to CPU0 and CPU1 PBX runs on HP ProLiant DL380 G5 server, OS is Gentoo Linux with 2.6.31 kernel. Other software versions: asterisk - 1.6.2.6 libpri - 1.4.10.2 dahdi - 2.2.0.2 any idea what could be the problem / what should I check to diagnose it ? Luke -- _ -- 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 and Patton
Hi! calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton) or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming from SIP/1004. Read up on how Asterisk does user/peer matching in sip.conf on inbound calls: With all users/peers having the same IP and hostname it is the entry that was defined last in sip.conf that wins. Here's a starter: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer Olle has often posted in more detail about this here. Either you simply do not differentiate between the different lines and treat them all as one single trunk (why exactly do you need to know which line is in use?), or you have to consider other ways like assigning different SIP ports on the Patton (a SIP gateway for each line), or maybe use different usernames when calling asterisk, or check if using different SIP domains (see [general] section in sip.conf) can help you. See also: http://www.mail-archive.com/asterisk-...@lists.digium.com/msg39355.html https://issues.asterisk.org/view.php?id=14340 https://issues.asterisk.org/view.php?id=14250 Note: Your issue is Patton -- Asterisk, while the registration part of the Patton config that you posted matters for Asterisk -- Patton calls. Philipp -- _ -- 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] Calls Dropping
i'm having the same problem with one of my call centers located in Egypt.. although the ip-phones are located on a Dedicated Leased Line yet calls drop out of the blue.almost an identical setup as yours..provider located in France (data center) my server located in Sweden (data center) both on public network no NAT.. and the remote office is behind NAT.somehow i suspect Internet problems with your case.. as RTP packets should not stop arriving unless internet connection is timing out. i suppose your calls that are dropping are INBOUND coming from your provider and directed to your remote location.. and you don't have any problems with OUTBOUND calls from your remote location to your server ( I have setup a loop test that goes between 5 locations originating from my remote location and returns to the remote location through 5 hops including IPKALL servers and call goes well with no problem). and let me take a wild guess.. your provider is offering a premium number services.my advise check your internet connection on the remote location and keep a ping from that network to your server running all the time to check for time outs. -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 From: d...@keshercommunications.com To: asterisk-users@lists.digium.com Date: Thu, 29 Apr 2010 16:33:06 -0400 Subject: [asterisk-users] Calls Dropping Hi, I’m having a major problem with random calls dropping. After spending weeks trying to figure it out, i’ve finally spotted the issue but don’t know how to resolve it. I run a sip server that’s hosted in a data centre. It has a public IP address with no nat involved. My provider also has a public ip with no nat involved. The sip phones are in a remote office behind a nat router. Every so often, all the rtp data coming from the remote location stops arriving at my sip server. So after about 30 seconds, the call gets terminated by my provider because i’m not sending any rtp packets to them. Any ideas why the rtp data should stop coming in, and how can I resolve it? Asterisk v1.4.30 6 x Linksys SPA921 Router at remote site is a Thomson TG585v7 Any assistance will be greatly appreciated. Many thanks Dan _ Hotmail is redefining busy with tools for the New Busy. Get more from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_2-- _ -- 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] GXW4024
Hello, I consider buying three GrandStream GXW4024 and connect 72 analogue phones to asterisk Do you have any feedback how well it works with Asterisk ? I am on a budget, do you have other recommendation for similar setup that get into same budget - connect around 70 analogue phones to asterisk. Thanks in advance. Peter -- _ -- 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] SpiderMux?
Tim Nelson wrote: Greetings all- I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks rather interesting. Has anyone used one? Where did you purchase it? Pricing? Operational issues? http://spidermux.com/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 A couple of things bother me about their webpage. The link for the manufacturers home page goes to an expired domain name. And the price list page is dated in 2006. -- _ -- 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 and Patton
On Fri, 30 Apr 2010 10:39:12 +0200, Carlo Dimaggio jaasmail...@gmail.com wrote: 2010/4/30 A.Santoro n...@ecoricerche.it Hi, we have and Asterisk server connected to a Patton Smartnode 4638 with 4 BRI. [...] Have you tried setting insecure=port,invite in the sip.conf for each sip account? Hi Carlo, thanks for your answer. Now I tried... and nothing is changed. In the following lines one the sip account of the peers (in sip.conf) [1001] username=1001 type=friend secret= dtmfmode=auto insecure=very host=dynamic port=5060 context=inbound qualify=yes disallow=all allow=ulaw allow=alaw canreinvite=no Bye. Eco -- _ -- 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] get hold event
Hi, all how to get hold event in asterisk. is it possible, when user1 put on hold in queue moh1 file played. when call transfer to agent and answered agent put hold at that time moh2 file played ? I have used asterisk 1.4 version. Regards, -- Bhrugu Mehta Sr. S/W Engineer (DD) VOIP,Telephony Team India -- _ -- 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] Confusion on call forwarding
Am 30.03.2010 20:56, schrieb Richard Kenner: You need promiscredir set to yes on sip.conf And then what do I do in the dialplan? I.e., what context is the redirect number interpreted in? Google searches on this issue show inconsistent and contradictory information. I usually set the context for transfers and forwards (3xx) manually. Set(__TRANSFER_CONTEXT=handleTransfer) Set(__FORWARD_CONTEXT=handleForward) regards klaus -- _ -- 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] PRI - Native ZAP bridge fails - Is this my patch?
The disconnect is RECEIVED by Asterisk. So there is a problem with the other party. You are sending FACILITY - maybe the other party does not like FACILITY and hangs up. IIRC there is a setting in zapata.conf to enable/disable FACILITY. regards klaus Am 10.04.2010 21:46, schrieb bruce bruce: Hi Guys, I am calling out 416-999- on Channel 1 of PRI and then calling 416-999- on Channel 2 of PRI. When the two channels are going to be ZAP native bridged, both channels hangup and CLI show PRI cause (16). Asterisk Verbose *(Channel 1 already connected to party)*: -- Requested transfer capability: 0x00 - SPEECH -- Called g0/416999 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap-bridge, s, 8) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Here is PRI debug, starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2967 q931_release: call 32864 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: RELEASE COMPLETE (90) q931.c:3766 q931_receive: call 32864 on channel 1 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Terminator) Message type: RELEASE (77) q931.c:3801 q931_receive: call 32865 on channel 2 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: RELEASE COMPLETE (90) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null System Info: *Bell Canada PRI* *Asterisk 1.4.21.2 * *Lib PRI 1.4.10* Is this my patch? https://issues.asterisk.org/view.php?id=7494 Thanks, Bruce --
[asterisk-users] Fwd: Re: SpiderMux?
Hi, I have one in stock - got it from a client who wanted to get rid of all his old IT equipment. Looks strange, did not have enough time to play with it Tried it once, looked hard to configure. It stays unused in the storage room. Peter On 29.4.2010 10:20, Tim Nelson wrote: Greetings all- I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks rather interesting. Has anyone used one? Where did you purchase it? Pricing? Operational issues? http://spidermux.com/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 -- _ -- 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 and Patton
On Fri, 30 Apr 2010 14:16:14 +0200, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton) or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming from SIP/1004. Read up on how Asterisk does user/peer matching in sip.conf on inbound calls: With all users/peers having the same IP and hostname it is the entry that was defined last in sip.conf that wins. Philipp thanks for your answer. This clears all my doubts, is not my configuration problem. why exactly do you need to know which line is in use? We have 4 trunk and 4 company in our office, I was testing FOP and I would want to show the occupied trunks for inbound and outbound calls for single company. Thanks again. Bye Eco -- _ -- 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] Fwd: Re: SpiderMux?
I have played with one before, it worked quite well. (Until somebody fried it by accident). Joachim Peter wrote: Hi, I have one in stock - got it from a client who wanted to get rid of all his old IT equipment. Looks strange, did not have enough time to play with it Tried it once, looked hard to configure. It stays unused in the storage room. Peter On 29.4.2010 10:20, Tim Nelson wrote: Greetings all- I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks rather interesting. Has anyone used one? Where did you purchase it? Pricing? Operational issues? http://spidermux.com/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 -- _ -- 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] PRI - Native ZAP bridge fails - Is this my patch?
Thanks. Yeah, that was the issue. I was requesting RLT and it wasn't turned ON with the provider. Your mentioned solution fixed it. -Bruce On Fri, Apr 30, 2010 at 9:59 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: The disconnect is RECEIVED by Asterisk. So there is a problem with the other party. You are sending FACILITY - maybe the other party does not like FACILITY and hangs up. IIRC there is a setting in zapata.conf to enable/disable FACILITY. regards klaus Am 10.04.2010 21:46, schrieb bruce bruce: Hi Guys, I am calling out 416-999- on Channel 1 of PRI and then calling 416-999- on Channel 2 of PRI. When the two channels are going to be ZAP native bridged, both channels hangup and CLI show PRI cause (16). Asterisk Verbose *(Channel 1 already connected to party)*: -- Requested transfer capability: 0x00 - SPEECH -- Called g0/416999 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap-bridge, s, 8) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Here is PRI debug, starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2967 q931_release: call 32864 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: RELEASE COMPLETE (90) q931.c:3766 q931_receive: call 32864 on channel 1 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Terminator) Message type: RELEASE (77) q931.c:3801 q931_receive: call 32865 on channel 2 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: RELEASE COMPLETE (90) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null System Info:
Re: [asterisk-users] GXW4024
On Fri, Apr 30, 2010 at 5:26 AM, Peter peterp...@aboutsupport.com wrote: I consider buying three GrandStream GXW4024 and connect 72 analogue phones to asterisk I recommend against that product. I have two that now sit on a shelf due to bad call quality, echo issues, and random one way audio... Do you have any feedback how well it works with Asterisk ? I am on a budget, do you have other recommendation for similar setup that get into same budget - connect around 70 analogue phones to asterisk. They are easy to setup and connect to Asterisk. That is about the only thing that they do well. I purchased two of these to try and fit within my budget, and ended up replacing them after about a month. The call quality was sub-par, and I had all kinds of echo issues. Firmware updates didn't seem to make anything better. I ended up replacing them with AudioCodes MP-124 which have been rock solid. Of course they cost about twice as much, but you get what you pay for. In the long run I went way over budget, but learned a good lesson! -Jonathan -- _ -- 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] Continuing after a TIMEOUT(absolute)
I don't think you are actually hitting the time out. Comment out the
set timeout line I think the results will be the same. Which tells me
the timeout is not kicking in.
On 4/29/10, Brendan Sterne bren...@callvine.com wrote:
Greetings,
I'm trying to continue to do some processing after a TIMEOUT
(absolute). In my dialplan below, when a call comes in to [default],
I call macro-phonenum and pass it a timeout of 20 seconds. macro-
phonenum sets TIMEOUT(absolute), then loops saying the phone number
that was called (in MACRO_EXTEN). When the timeout expires I want to
call my macro-hangup (so it can say goodbye or whatever). But the
system is just hanging up. The dialplan and log output is below. Any
info is appreciated. This is on version 1.6.0.5.
[macro-answer-and-join]
exten = s,1,NoOp()
exten = s,n,Answer()
exten = s,n,Wait(4)
exten = s,n,SendDTMF(1)
exten = s,n,Wait(1)
exten = s,n,SendDTMF(1)
exten = s,n,MacroExit
[macro-hangup]
exten = s,1,NoOp()
exten = s,n,Playback(goodbye)
exten = s,n,Hangup()
;
exten = T,1,NoOp()
exten = T,n,Playback(goodbye)
exten = T,n,Hangup()
[macro-phonenum]
exten = s,1,NoOp()
exten = s,n,Macro(answer-and-join)
exten = s,n,Set(TIMEOUT(absolute)=${ARG1})
exten = s,n,Set(i=1000)
exten = s,n,While($[${i} = 1])
exten = s,n,SayDigits(${MACRO_EXTEN})
exten = s,n,Wait(5)
exten = s,n,Set(i=$[${i} - 1])
exten = s,n,EndWhile()
exten = s,n,MacroExit
;
exten = T,1,NoOp()
exten = T,n,Macro(hangup)
exten = T,n,MacroExit
[default]
exten = _X.,1,NoOp()
exten = _X.,n,Macro(phonenum,20)
exten = _X.,n,Macro(hangup)
;
exten = T,1,NoOp()
exten = T,n,Macro(hangup)
The log when the timeout occurs:
snip (I'm in macro-phonenum)
-- SIP/70.124.61.17-082a69a8 Playing 'digits/5.ulaw' (language
'en')
-- SIP/70.124.61.17-082a69a8 Playing 'digits/1.ulaw' (language
'en')
-- SIP/70.124.61.17-082a69a8 Playing 'digits/2.ulaw' (language
'en')
-- SIP/70.124.61.17-082a69a8 Playing 'digits/1.ulaw' (language
'en')
-- SIP/70.124.61.17-082a69a8 Playing 'digits/2.ulaw' (language
'en')
-- Executing [...@macro-phonenum:7] Wait(SIP/
70.124.61.17-082a69a8, 5) in new stack
== Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/
70.124.61.17-082a69a8' in macro 'phonenum'
== Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/
70.124.61.17-082a69a8'
Scheduling destruction of SIP dialog 'D8FE9724-1DD1-11B2-9F1A-
a4ef9db84...@192.168.1.98' in 32000 ms (Method: ACK)
set_destination: Parsing sip:70.124.61.17:5060 for address/port to
send to
set_destination: set destination to 70.124.61.17, port 5060
Reliably Transmitting (NAT) to 70.124.61.17:5060:
BYE sip:70.124.61.17:5060 SIP/2.0
snip
Cheers,
- Brendan
Brendan Sterne
QA Lead, Callvine
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Re: [asterisk-users] Call-Waiting, implementation ideas
If you use zap then asterisk already does it. With sip the phones will not tell asterisk about the hook flash. However you can play around with dynamic features and assign a key that will mimic hook flash. Injecting the beep sound might be hard though. Playing a different ring to 2nd caller based on if the recipient is on the phone can be accomplished using chanavail or whatever that app is called can't recall at the moment and I'm typing this on my BB On 4/30/10, Harel Cohen ha...@easycall.gi wrote: Hi all, How can I implement a full-featured Call-Waiting behavior on the Asterisk level (e.g. I don't want to relay on end-equipment capabilities)? I found it very strange that such a basic feature is not built-in in Asterisk (and I've googled a lot in search for this). Here is what I need: SomeuserX is calling MyUserA. They are on conversation (assumption: voice is via the Asterisk) SomeuserY is calling MyUserA. SomeuserY gets a special ringing tone. Meaning - Asterisk opens voice channel towards SomeuserY (progress with SDP) and plays SpecialRingBack.wav/gsm etc. MyUserA Gets voice notification (e.g. beep-beep) during his call to SomeuserX. Meaning - Asterisk barge-in the rtp stream and play the file beepbeep.wav/gsm on the MyUserA channel. This is done periodically for as long as SomeuserY is waiting to be answered (i.e. doesn't hang-up). Asterisk is monitoring the state of the call SomeuserX - MyUserA. If MyUserA will signal (e.g. hook-flash or some digit sequence) that he wants to answer the 2nd call then Asterisk will put on hold SomeuserX and bridge SomeuserY to MyUserA with the option for MyUserA to toggle between the two channels. If the conversation SomeuserX with MyUserA is terminated Asterisk will INVITE MyUserA and when picked up will bridge SomeuserY with MyUserA. I hope there is a solution for that… I tried using DEVICE_STATE for this purpose however I keep getting status NOT_INUSE even if the extension IS in use (I'll open a different thread on this issue if needed). Thanks in advance for any ideas provided, Harel -- _ -- 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 stopping for no reason
2010/4/30 Motiejus Jakštys desired@gmail.com Hi, please always add asterisk version to your query. I am using Asterisk 1.4.17~dfsg-2ubuntu1.1 on an Ubuntu 8.04.4 server. I managed to run internet radio (that streams MP3) within asterisk. Minor change is nescesarry to make it work with random MP3s. My Dialplan: exten = _X.,n,Answer() exten = _X.,n,MP3Player(http://stream.m-1.fm/m1/mp3) $ cat /usr/bin/mpg123 #!/bin/bash /usr/bin/wget -q -O - $1 | /usr/bin/madplay -Q -z -o raw:- --mono -R 8000 -a -6 - You should change the WGET part to something that better suits your needs. Tested on asterisk 1.4.27 Good. Is there a way to set random in files mode? -- _ -- 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] Issue with (pattern) matching extension
Andrew Latham wrote: Are you guys talking about the Asterisk Cookbook Because that could be released in the next 20 years at this point... The Asterisk Cookbook probably won't ever be released unless someone else wants to step up and start it. We (as in the authors of Asterisk: TFoT) had some discussions about doing the book, but it was released as a real book before we actually signed a contract, and we have since gotten to busy to start a new book from scratch. However, if someone wants to write a book, I'm sure O'Reilly would be happy to hear from you about picking it up and running with it. Leif. -- _ -- 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] B400P card crashes conncection
Hi, I have a B400P BRI card with point-to-point connection (signalling: bri_cpe) with this dmesg: http://pastebin.com/sXrRt1yM When i restart asterisk server, the card cannot connect to the telco, the control led flashes red. If I unplug the cable between the ISDN nt and the card and wait 40 sec, the card can connect and works properly. The telco says the asterisk crashes the connection with the telco, when I let the NT reconnect, the card connects properly. Do you have any idea how to solve this problem? Thanks for any help in advance. Best regards, Peter Gelencser -- _ -- 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] SpiderMux?
- Lyle Giese l...@lcrcomputer.net wrote: Tim Nelson wrote: Greetings all- I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks rather interesting. Has anyone used one? Where did you purchase it? Pricing? Operational issues? http://spidermux.com/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 A couple of things bother me about their webpage. The link for the manufacturers home page goes to an expired domain name. And the price list page is dated in 2006. I agree... the pages and information are quite out of date. However, I can't seem to find any place that sells these units. I'm not in dire need of one, I simply found the product and found it to be very interesting to say the least. The only other device I've used which brings the calls into Asterisk via TDMoE is the Fonebridge which worked very well. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 -- _ -- 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] Continuing after a TIMEOUT(absolute)
CF,
When I comment out the timeout the call continues as expected. I
believe the timeout is kicking in.
Can anyone point me to an example where TIMEOUT(absolute) is used as a
general timer, where the call continues after the expiry? I'm not
sure which extension to use T or t. I've tried both but neither
seem to work.
Cheers,
- Brendan
Brendan Sterne
QA Lead, Callvine
On Apr 30, 2010, at 9:38 AM, C F wrote:
I don't think you are actually hitting the time out. Comment out the
set timeout line I think the results will be the same. Which tells me
the timeout is not kicking in.
On 4/29/10, Brendan Sterne bren...@callvine.com wrote:
Greetings,
I'm trying to continue to do some processing after a TIMEOUT
(absolute). In my dialplan below, when a call comes in to [default],
I call macro-phonenum and pass it a timeout of 20 seconds. macro-
phonenum sets TIMEOUT(absolute), then loops saying the phone number
that was called (in MACRO_EXTEN). When the timeout expires I want to
call my macro-hangup (so it can say goodbye or whatever). But the
system is just hanging up. The dialplan and log output is below.
Any
info is appreciated. This is on version 1.6.0.5.
[macro-answer-and-join]
exten = s,1,NoOp()
exten = s,n,Answer()
exten = s,n,Wait(4)
exten = s,n,SendDTMF(1)
exten = s,n,Wait(1)
exten = s,n,SendDTMF(1)
exten = s,n,MacroExit
[macro-hangup]
exten = s,1,NoOp()
exten = s,n,Playback(goodbye)
exten = s,n,Hangup()
;
exten = T,1,NoOp()
exten = T,n,Playback(goodbye)
exten = T,n,Hangup()
[macro-phonenum]
exten = s,1,NoOp()
exten = s,n,Macro(answer-and-join)
exten = s,n,Set(TIMEOUT(absolute)=${ARG1})
exten = s,n,Set(i=1000)
exten = s,n,While($[${i} = 1])
exten = s,n,SayDigits(${MACRO_EXTEN})
exten = s,n,Wait(5)
exten = s,n,Set(i=$[${i} - 1])
exten = s,n,EndWhile()
exten = s,n,MacroExit
;
exten = T,1,NoOp()
exten = T,n,Macro(hangup)
exten = T,n,MacroExit
[default]
exten = _X.,1,NoOp()
exten = _X.,n,Macro(phonenum,20)
exten = _X.,n,Macro(hangup)
;
exten = T,1,NoOp()
exten = T,n,Macro(hangup)
The log when the timeout occurs:
snip (I'm in macro-phonenum)
-- SIP/70.124.61.17-082a69a8 Playing 'digits/5.ulaw' (language
'en')
-- SIP/70.124.61.17-082a69a8 Playing 'digits/1.ulaw' (language
'en')
-- SIP/70.124.61.17-082a69a8 Playing 'digits/2.ulaw' (language
'en')
-- SIP/70.124.61.17-082a69a8 Playing 'digits/1.ulaw' (language
'en')
-- SIP/70.124.61.17-082a69a8 Playing 'digits/2.ulaw' (language
'en')
-- Executing [...@macro-phonenum:7] Wait(SIP/
70.124.61.17-082a69a8, 5) in new stack
== Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/
70.124.61.17-082a69a8' in macro 'phonenum'
== Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/
70.124.61.17-082a69a8'
Scheduling destruction of SIP dialog 'D8FE9724-1DD1-11B2-9F1A-
a4ef9db84...@192.168.1.98' in 32000 ms (Method: ACK)
set_destination: Parsing sip:70.124.61.17:5060 for address/port to
send to
set_destination: set destination to 70.124.61.17, port 5060
Reliably Transmitting (NAT) to 70.124.61.17:5060:
BYE sip:70.124.61.17:5060 SIP/2.0
snip
Cheers,
- Brendan
Brendan Sterne
QA Lead, Callvine
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Re: [asterisk-users] Asterisk and Patton
Hi! This clears all my doubts, is not my configuration problem. As I said, you could think about creating 4 different SIP gateways on the Patton with 4 differing SIP ports. I don't know if the Patton will handle 4 gateways - but it might. We have 4 trunk and 4 company in our office, I was testing FOP and I would want to show the occupied trunks for inbound and outbound calls for single company. Alternatives are: - use GROUP() and GROUP_COUNT in the dialplan - use DEVICE_STATE in the dialplan This includes a lenghty example on how to monitor a BRI trunk: http://www.voip-info.org/wiki/view/Asterisk+func+device_State Philipp -- _ -- 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 and Patton
2010/4/30 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de Hi! This clears all my doubts, is not my configuration problem. As I said, you could think about creating 4 different SIP gateways on the Patton with 4 differing SIP ports. I don't know if the Patton will handle 4 gateways - but it might. It does (at least with smartware 5.3). -- _ -- 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] Embedded IAX
Hi All, I've been lurking here for a while now, having only made a couple of posts. I am starting a new hardphone project and was wondering if there is some GPL'ed IAX source that I could start with. I've searched and haven't come up with much beyond iaxClient. While iaxClient does give me a little bit to start with, it looks like it is really intended to be more of a softphone running on a Linux machine, and will take some heavy mods to get it running in an embedded DSP environment. Running something like AstLinux on the DSP along with iaxClient may be a possibility but it seems like an awfully lot of baggage to carry around just to get the IAX part of the project. Any pointers would be greatly appreciated. Best, Bill -- _ -- 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] Follow-me to my answering machine :-(
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Bryan Jacobs wrote: I wonder if all the cell providers let you do this? I presume you mean turn off voice mail. I don't know, but the first time I called Verizon to have it done the gal I spoke with said it couldn't be done. So I said thanks, called in again, got another rep and he said no problem. In less than five minutes I was good to go. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFL2yNWCFu3bIiwtTARArEgAJ9TMJK0qgu/GkapCgjK+zPT+crHaACfQ03X BbTtSecEA2Ahuiqwws+2l10= =hjFW -END PGP SIGNATURE- -- _ -- 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] Follow-me to my answering machine :-(
On Fri, 30 Apr 2010, Barry L. Kline wrote: Bryan Jacobs wrote: I wonder if all the cell providers let you do this? I presume you mean turn off voice mail. I don't know, but the first time I called Verizon to have it done the gal I spoke with said it couldn't be done. So I said thanks, called in again, got another rep and he said no problem. In less than five minutes I was good to go. I have a t-mobile sidekick. I just found the menu where I set the voicemail phone number and changed it to my * box. I could've left it blank for no transfer at all. Vince. -- Michigan VHF Corp. http://www.nobucks.net/ http://www.CDupe.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] Embedded IAX
http://downloads.asterisk.org/pub/telephony/libiax/ That package is outdated AFAIK but is a start. You should be able to use chan_iax in Asterisk as a reference to fix libiax and use it for your own purposes. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com On Fri, Apr 30, 2010 at 1:23 PM, Bill Shaw b.s...@comcast.net wrote: Hi All, I've been lurking here for a while now, having only made a couple of posts. I am starting a new hardphone project and was wondering if there is some GPL'ed IAX source that I could start with. I've searched and haven't come up with much beyond iaxClient. While iaxClient does give me a little bit to start with, it looks like it is really intended to be more of a softphone running on a Linux machine, and will take some heavy mods to get it running in an embedded DSP environment. Running something like AstLinux on the DSP along with iaxClient may be a possibility but it seems like an awfully lot of baggage to carry around just to get the IAX part of the project. Any pointers would be greatly appreciated. Best, Bill -- _ -- 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] Strange Invite issue
Before posting let me mention that this doesn't happen with ALL destination on this provider.. some destination doesn't face this problem .. but this is a sample call [K -- Executing [0020100324...@a2billing:1] [1;36;40mDeadAGI[0;37;40m([1;35;40mSIP/58169-ac47fda0[0;37;40m, [1;35;40ma2billing.php|1[0;37;40m) in new stack [K -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- AGI Script Executing Application: (Dial) Options: (SIP/PROVIDER1/20100324519|60|HL(166986000:61000:3)) -- Limit Data for this call: timelimit = 166986000 play_warning = 61000 play_to_caller = yes play_to_callee = no warning_freq = 3 start_sound = (null) warning_sound = timeleft end_sound = (null)Audio is at 100.X.Y.Z port 13984Adding codec 0x100 (g729) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 195.X.Y.Z:5060:INVITE sip:20100324...@195.x.y.z SIP/2.0 Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport From: 58169 sip:58...@100.x.y.z;tag=as00522e07 To: sip:20100324...@195.x.y.z Contact: sip:58...@100.x.y.z Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 30 Apr 2010 18:52:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 267 v=0 o=root 12516 12516 IN IP4 100.X.Y.Z s=session c=IN IP4 100.X.Y.Z t=0 0 m=audio 13984 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 --- -- Called PROVIDER1/20100324519 [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 100 Trying Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 sip:58...@100.x.y.z;tag=as00522e07 To: sip:20100324...@195.x.y.z;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Content-Length: 0 - [K --- (7 headers 0 lines) --- [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 180 Ringing Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 sip:58...@100.x.y.z;tag=as00522e07 To: sip:20100324...@195.x.y.z;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Contact: sip:20100324...@195.x.y.z:5060 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 260 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 10183 6645 IN IP4 195.X.Y.Z s=SIP Media Capabilities c=IN IP4 195.219.240.5 t=0 0 m=audio 15846 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly a=maxptime:20 - [K --- (11 headers 12 lines) --- [K Found RTP audio format 18 [K Found RTP audio format 101 [K Peer audio RTP is at port 195.219.240.5:15846 [K Found audio description format G729 for ID 18 [K Found audio description format telephone-event for ID 101 [K Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) [K Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [K Peer audio RTP is at port 195.219.240.5:15846 [K -- SIP/PROVIDER1-1fd586a0 is ringing [K -- Call on SIP/PROVIDER1-1fd586a0 placed on hold [K -- Started music on hold, class 'default', on SIP/58169-ac47fda0 [K -- SIP/PROVIDER1-1fd586a0 is making progress passing it to SIP/58169-ac47fda0 [K sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 195.X.Y.Z 2010032451 7f169cce700 00102/0 0x100 (g729) Yes Init: INVITE 78.184.197.119 58169 AC8455D8edd 00101/160518 0x4 (ulaw) No Rx: INVITE 2 active SIP channels [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 180 Ringing Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 sip:58...@100.x.y.z;tag=as00522e07 To: sip:20100324...@195.x.y.z;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Contact: sip:20100324...@195.x.y.z:5060 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 0 - [K --- (9 headers 0 lines) --- [K -- SIP/PROVIDER1-1fd586a0 is ringing -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 Date: Thu, 29 Apr 2010 16:52:24 +0100 From: list-aster...@skycomuk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Strange Invite issue Can you post a sip debug Tarek Sawah wrote: Greetings List. I'm facing a strange issue with one of my
Re: [asterisk-users] Strange Invite issue
in the SIP/2.0 180 Ringing, the SDP shows: a=sendonly this is hold by rfc 3264. then when the other end picks up, a new SDP is probably sent with a=sendrecv I believe your server is acting correctly. -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Tarek Sawah Sent: Fri 4/30/2010 12:11 PM To: Asterisk Users Subject: Re: [asterisk-users] Strange Invite issue Before posting let me mention that this doesn't happen with ALL destination on this provider.. some destination doesn't face this problem .. but this is a sample call [K -- Executing [0020100324...@a2billing:1] [1;36;40mDeadAGI[0;37;40m([1;35;40mSIP/58169-ac47fda0[0;37;40m, [1;35;40ma2billing.php|1[0;37;40m) in new stack [K -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- AGI Script Executing Application: (Dial) Options: (SIP/PROVIDER1/20100324519|60|HL(166986000:61000:3)) -- Limit Data for this call: timelimit = 166986000 play_warning = 61000 play_to_caller = yes play_to_callee = no warning_freq = 3 start_sound = (null) warning_sound = timeleft end_sound = (null)Audio is at 100.X.Y.Z port 13984Adding codec 0x100 (g729) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 195.X.Y.Z:5060:INVITE sip:20100324...@195.x.y.z SIP/2.0 Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport From: 58169 sip:58...@100.x.y.z;tag=as00522e07 To: sip:20100324...@195.x.y.z Contact: sip:58...@100.x.y.z Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 30 Apr 2010 18:52:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 267 v=0 o=root 12516 12516 IN IP4 100.X.Y.Z s=session c=IN IP4 100.X.Y.Z t=0 0 m=audio 13984 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 --- -- Called PROVIDER1/20100324519 [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 100 Trying Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 sip:58...@100.x.y.z;tag=as00522e07 To: sip:20100324...@195.x.y.z;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Content-Length: 0 - [K --- (7 headers 0 lines) --- [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 180 Ringing Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 sip:58...@100.x.y.z;tag=as00522e07 To: sip:20100324...@195.x.y.z;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Contact: sip:20100324...@195.x.y.z:5060 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 260 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 10183 6645 IN IP4 195.X.Y.Z s=SIP Media Capabilities c=IN IP4 195.219.240.5 t=0 0 m=audio 15846 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly a=maxptime:20 - [K --- (11 headers 12 lines) --- [K Found RTP audio format 18 [K Found RTP audio format 101 [K Peer audio RTP is at port 195.219.240.5:15846 [K Found audio description format G729 for ID 18 [K Found audio description format telephone-event for ID 101 [K Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) [K Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [K Peer audio RTP is at port 195.219.240.5:15846 [K -- SIP/PROVIDER1-1fd586a0 is ringing [K -- Call on SIP/PROVIDER1-1fd586a0 placed on hold [K -- Started music on hold, class 'default', on SIP/58169-ac47fda0 [K -- SIP/PROVIDER1-1fd586a0 is making progress passing it to SIP/58169-ac47fda0 [K sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 195.X.Y.Z 2010032451 7f169cce700 00102/0 0x100 (g729) Yes Init: INVITE 78.184.197.119 58169 AC8455D8edd 00101/160518 0x4 (ulaw) No Rx: INVITE 2 active SIP channels [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 180 Ringing Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 sip:58...@100.x.y.z;tag=as00522e07 To: sip:20100324...@195.x.y.z;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Contact: sip:20100324...@195.x.y.z:5060 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 0 - [K --- (9 headers 0 lines) --- [K --
Re: [asterisk-users] Strange Invite issue
then why is it happening on a few destinations on that particular provider? Date: Fri, 30 Apr 2010 13:09:05 -0700 From: david.wh...@watchguard.com To: asterisk-users@lists.digium.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Strange Invite issue in the SIP/2.0 180 Ringing, the SDP shows: a=sendonly this is hold by rfc 3264. then when the other end picks up, a new SDP is probably sent with a=sendrecv I believe your server is acting correctly. -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Tarek Sawah Sent: Fri 4/30/2010 12:11 PM To: Asterisk Users Subject: Re: [asterisk-users] Strange Invite issue Before posting let me mention that this doesn't happen with ALL destination on this provider.. some destination doesn't face this problem .. but this is a sample call [K -- Executing [0020100324...@a2billing:1] [1;36;40mDeadAGI[0;37;40m([1;35;40mSIP/58169-ac47fda0[0;37;40m, [1;35;40ma2billing.php|1[0;37;40m) in new stack [K -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- AGI Script Executing Application: (Dial) Options: (SIP/PROVIDER1/20100324519|60|HL(166986000:61000:3)) -- Limit Data for this call: timelimit = 166986000 play_warning = 61000 play_to_caller = yes play_to_callee = no warning_freq = 3 start_sound = (null) warning_sound = timeleft end_sound = (null)Audio is at 100.X.Y.Z port 13984Adding codec 0x100 (g729) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 195.X.Y.Z:5060:INVITE sip:20100324...@195.x.y.z SIP/2.0 Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport From: 58169 ;tag=as00522e07 To: Contact: Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 30 Apr 2010 18:52:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 267 v=0 o=root 12516 12516 IN IP4 100.X.Y.Z s=session c=IN IP4 100.X.Y.Z t=0 0 m=audio 13984 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 --- -- Called PROVIDER1/20100324519 [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 100 Trying Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 ;tag=as00522e07 To: ;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Content-Length: 0 - [K --- (7 headers 0 lines) --- [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 180 Ringing Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 ;tag=as00522e07 To: ;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Contact: Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 260 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 10183 6645 IN IP4 195.X.Y.Z s=SIP Media Capabilities c=IN IP4 195.219.240.5 t=0 0 m=audio 15846 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly a=maxptime:20 - [K --- (11 headers 12 lines) --- [K Found RTP audio format 18 [K Found RTP audio format 101 [K Peer audio RTP is at port 195.219.240.5:15846 [K Found audio description format G729 for ID 18 [K Found audio description format telephone-event for ID 101 [K Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) [K Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [K Peer audio RTP is at port 195.219.240.5:15846 [K -- SIP/PROVIDER1-1fd586a0 is ringing [K -- Call on SIP/PROVIDER1-1fd586a0 placed on hold [K -- Started music on hold, class 'default', on SIP/58169-ac47fda0 [K -- SIP/PROVIDER1-1fd586a0 is making progress passing it to SIP/58169-ac47fda0 [K sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 195.X.Y.Z 2010032451 7f169cce700 00102/0 0x100 (g729) Yes Init: INVITE 78.184.197.119 58169 AC8455D8edd 00101/160518 0x4 (ulaw) No Rx: INVITE 2 active SIP channels [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 180 Ringing Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 ;tag=as00522e07 To: ;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Contact: Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length:
Re: [asterisk-users] Strange Invite issue
I don't know in your particular case, but if I call a PSTN endpoint via my provider, the SIP signaling is different than if I'm calling a remote SIP endpoint. This is because PSTN gateways have to make decisions (about codecs, eg) independently of the remote endpoints. In other words, remote SIP endpoints generate their own SDPs, which your provider forwards to you. Gateways often have to generate their own. Those SDPs will necessarily be different. -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Tarek Sawah Sent: Fri 4/30/2010 2:49 PM To: Asterisk Users Subject: Re: [asterisk-users] Strange Invite issue then why is it happening on a few destinations on that particular provider? Date: Fri, 30 Apr 2010 13:09:05 -0700 From: david.wh...@watchguard.com To: asterisk-users@lists.digium.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Strange Invite issue in the SIP/2.0 180 Ringing, the SDP shows: a=sendonly this is hold by rfc 3264. then when the other end picks up, a new SDP is probably sent with a=sendrecv I believe your server is acting correctly. -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Tarek Sawah Sent: Fri 4/30/2010 12:11 PM To: Asterisk Users Subject: Re: [asterisk-users] Strange Invite issue Before posting let me mention that this doesn't happen with ALL destination on this provider.. some destination doesn't face this problem .. but this is a sample call [K -- Executing [0020100324...@a2billing:1] [1;36;40mDeadAGI[0;37;40m([1;35;40mSIP/58169-ac47fda0[0;37;40m, [1;35;40ma2billing.php|1[0;37;40m) in new stack [K -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- AGI Script Executing Application: (Dial) Options: (SIP/PROVIDER1/20100324519|60|HL(166986000:61000:3)) -- Limit Data for this call: timelimit = 166986000 play_warning = 61000 play_to_caller = yes play_to_callee = no warning_freq = 3 start_sound = (null) warning_sound = timeleft end_sound = (null)Audio is at 100.X.Y.Z port 13984Adding codec 0x100 (g729) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 195.X.Y.Z:5060:INVITE sip:20100324...@195.x.y.z SIP/2.0 Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport From: 58169 ;tag=as00522e07 To: Contact: Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 30 Apr 2010 18:52:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 267 v=0 o=root 12516 12516 IN IP4 100.X.Y.Z s=session c=IN IP4 100.X.Y.Z t=0 0 m=audio 13984 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 --- -- Called PROVIDER1/20100324519 [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 100 Trying Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 ;tag=as00522e07 To: ;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Content-Length: 0 - [K --- (7 headers 0 lines) --- [K --- SIP read from 195.X.Y.Z:5060 ---SIP/2.0 180 Ringing Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060 From: 58169 ;tag=as00522e07 To: ;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Contact: Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 260 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 10183 6645 IN IP4 195.X.Y.Z s=SIP Media Capabilities c=IN IP4 195.219.240.5 t=0 0 m=audio 15846 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly a=maxptime:20 - [K --- (11 headers 12 lines) --- [K Found RTP audio format 18 [K Found RTP audio format 101 [K Peer audio RTP is at port 195.219.240.5:15846 [K Found audio description format G729 for ID 18 [K Found audio description format telephone-event for ID 101 [K Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) [K Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [K Peer audio RTP is at port 195.219.240.5:15846 [K -- SIP/PROVIDER1-1fd586a0 is ringing [K -- Call on SIP/PROVIDER1-1fd586a0 placed on hold [K -- Started music on hold, class 'default', on SIP/58169-ac47fda0 [K -- SIP/PROVIDER1-1fd586a0 is making progress passing it to
Re: [asterisk-users] AGI == DeadAGI
It is irrelevant who hangs up, you want to just use DeadAGI in the h extension I wish that would be the case, but at least on 1.4 I see: [Apr 30 14:59:38] -- Executing [...@master-route:1] DeadAGI(...) in new stack [Apr 30 14:59:38] WARNING[27845]: res_agi.c:2160 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI The good news is, we run tens of thousands of calls every day through this box and about half of them spit out this warning, but it never caused any problems for over a year. Thus this warning is probably safe to ignore. Luki -- _ -- 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 change in payload. (SDP)
Thanks a lot Kevin for the reply From: Kevin P. Fleming kpflem...@digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, April 29, 2010 5:43:15 AM Subject: Re: [asterisk-users] No change in payload. (SDP) Aditya Kumar wrote: re-posting the question. --- use case: when some one in my pbx calls 100.200, I have translations well defined, Media also (media via asterisk) --Works. when some one calls bob, or for any names I am adding Domain and call is been sent to the other party -- Works, no media... For the cases when it is talking to the external work, I want Astersik not to do anything with the SDP/payload. I want it to send as it is to the external proxy. How can I achieve this? so that the SDP/payload will not be modified for users talking to the external world. I want media for those external devices to come Directly to the users in my pbx. (with out going t asterisk) 2) also related question is can I have the xml payload in the originator and call is routed via PBX to the Target. The xml payload also must be carried to the target. is it possible This will really help me as I was held up with this :( Neither of these are possible; Asterisk is a B2BUA, not a proxy, and as such the outgoing INVITE is a *different* session from the incoming one. That means that Asterisk has to be able to understand the SDP content that arrives so it can forward media between the two sessions. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
