Re: [asterisk-users] voipmonitor.org
On 8.5.2010 00:40, Jeff Brower wrote: Martin- checkout new open source voipmonitor.org SIP packet sniffer. I've developed it for my telco company and I've decided to share it. Testing and contributions are welcome! VoIPmonitor is open source live network packet sniffer which analyze SIP and RTP protocol. It can run as daemon or analyzes already captured pcap files. For each detected VoIP call voipmonitor calculates statistics about loss, burstiness, latency and predicts MOS (Meaning Opinion Score) according to ITU-T G.107 E-model. These statistics are saved to MySQL database and each call is saved as pcap dump. Web PHP application (it is not part of open source sniffer) filters data from database and graphs latency and loss distribution. Voipmonitor also detects improperly terminated calls when BYE or OK was not seen. To accuratly transform latency to loss packets, voipmonitor simulates fixed and adaptive jitterbuffer. How many channels can it handle simultaneously? I've not tested limits but capturing 15 voip calls takes 3-4% on Core2 2.40GHz. Complexity in worst case is O(N^2) where N is number of calls. Packets are matched as llinear list of IP and port. If this will be limit, it could be rewriten to hash table O(N) How does it do MOS prediction if low bitrate codecs are being used (G729, AMR, etc)? It is calibrated only to G.711 with PLC for now but I'm planing adding equations for G.729 and iLBC. MV -- _ -- 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] voipmonitor.org
On Mon, May 10, 2010 at 1:09 PM, Martin Vít v...@lam.cz wrote: On 8.5.2010 00:40, Jeff Brower wrote: Martin- checkout new open source voipmonitor.org SIP packet sniffer. I've developed it for my telco company and I've decided to share it. Testing and contributions are welcome! VoIPmonitor is open source live network packet sniffer which analyze SIP and RTP protocol. It can run as daemon or analyzes already captured pcap files. For each detected VoIP call voipmonitor calculates statistics about loss, burstiness, latency and predicts MOS (Meaning Opinion Score) according to ITU-T G.107 E-model. These statistics are saved to MySQL database and each call is saved as pcap dump. Web PHP application (it is not part of open source sniffer) filters data from database and graphs latency and loss distribution. Voipmonitor also detects improperly terminated calls when BYE or OK was not seen. To accuratly transform latency to loss packets, voipmonitor simulates fixed and adaptive jitterbuffer. How many channels can it handle simultaneously? I've not tested limits but capturing 15 voip calls takes 3-4% on Core2 2.40GHz. Complexity in worst case is O(N^2) where N is number of calls. Packets are matched as llinear list of IP and port. If this will be limit, it could be rewriten to hash table O(N) How does it do MOS prediction if low bitrate codecs are being used (G729, AMR, etc)? It is calibrated only to G.711 with PLC for now but I'm planing adding equations for G.729 and iLBC. when are you expecting to release Ram -- _ -- 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] dialplan question
Hello. Look, in Your sip.conf [601] deny=0.0.0.0/0.0.0.0 - you deny all, but some lines down you allow from all context=office type=friend secret=601 qualify=yes ;port=5060 permit=0.0.0.0/0.0.0.0 nat=no mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/601 canreinvite=no callgroup=1 pickupgroup=1 callerid=device601 accountcode= - delete this line call-limit=50 Vasiliy G Tolstov wrote: Hello. I'm new with asterisk. Can you help me in this: I have cisco sip phone (601) connected to asterisk server, and 1 client number (500). I want to dial from 601 to 500. But get error in cli console: [Apr 27 15:30:15] NOTICE[9650]: chan_sip.c:20059 handle_request_invite: Call from '601' to extension '500' rejected because extension not found. What's wrong? extensions.conf: [office] exten = 601,1,Answer() exten = 601,2,Wait,2 exten = 601,3,Dial(SIP/601,20) exten = 601,4,Hangup() exten = 500,1,Answer() exten = 500,2,Wait,2 exten = 500,3,Dial(SIP/500,20) exten = 500,4,Hangup() sip.conf: [601] deny=0.0.0.0/0.0.0.0 context=office type=friend secret=601 qualify=yes ;port=5060 permit=0.0.0.0/0.0.0.0 nat=no mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/601 canreinvite=no callgroup=1 pickupgroup=1 callerid=device601 accountcode= call-limit=50 [500] deny=0.0.0.0/0.0.0.0 username=500 context=office type=friend secret=500 qualify=yes ;port=5060 permit=0.0.0.0/0.0.0.0 nat=no mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/500 canreinvite=no callgroup=1 pickupgroup=1 callerid=device500 accountcode= call-limit=50 -- _ -- 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] Problem of Playing 'pbx-transfer'
Try x-lite to x-lite, snom to snom . That may be a codec problem. Which codec are you using? Adolphe Cher-aime From my Iphone On May 9, 2010, at 11:11 PM, Dovid Bender asteriskus...@dovid.net wrote: Process of elemination. Test with multiple phones, check the codec being used and make sure the file is there and available. - Original Message - From: kamrun nahar bina To: asterisk-users@lists.digium.com Sent: Friday, May 07, 2010 07:33 Subject: [asterisk-users] Problem of Playing 'pbx-transfer' Dear all, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer. During attended transfer, sometimes we cannot hear the sound of 'pbx- transfer'. I cannot understand why this is happening? log is : -- Started music on hold, class 'default', on SIP/113.34.235.13- b7a3f110 -- SIP/185148-092db338 Playing 'pbx-transfer' (language 'jp') Although it is showing Playing 'pbx-transfer' (language 'jp'), but it cannot hear 'pbx-transfer' sound Sometimes we can hear the sound of 'pbx-transfer'. is it the problem of network load or phone-set or something else? Please let me know. I am using x-lite and snom 300. Before i tested it for memory load, And found out that it is not a memory problem. Our system is as like as: The number of User agent is: 1650 The number of Actual registered user agent is: 600 Our System configuration is : IBM X3550 CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz HDD: 3.5 SATA 1TB x 2 version of asterisk: 1.4.23.1 our memory size is 4GB. concurrent calls no : 30. Our memory condition is below : Cpu(s): 0.3%us, 0.7%sy, 0.0%ni, 98.5%id, 0.0%wa, 0.1%hi, 0.3%si, 0.0%st Mem: 4147888k total, 3986540k used, 161348k free,76852k buffers Swap: 2031608k total, 56k used, 2031552k free, 3170396k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 23160 root 15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asterisk Our disk space condition is below: FilesystemSize Used Avail Use% Mounted on /dev/mapper/VolGroup00-LogVol00 901G 245G 610G 29% / /dev/sda1 99M 18M 77M 19% /boot tmpfs 2.0G 0 2.0G 0% /dev/shm Asterisk and the User-Agent is connected through the Internet. ..And Is there any solution to solve this problem? I have investigated in several places but I cannot find out the reason? I need this solution very urgently. Is there any one who can solve this problem? -- _ -- 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
[asterisk-users] Simulating a commercial SIP provider
Hi all, The kind of configuration that I use in my sip.conf to connect to various commercial SIP providers looks like this: [general] context=incoming-calls canreinvite=no qualify=yes register = jwinius:pass...@sip.provider.com/0201234567 [provider] type=peer host=sip.provider.com fromuser=jwinius secret=passwrd This works. However, how would I have to configure the sip.conf of a second Asterisk machine if I wanted to use it to simulate the host mentioned above, sip.provider.com, but (crucially) without changing the above configuration? I would have thought that the appropriate stanza to use for my account in the other Asterisk machine's sip.conf -- the system that simulates the commercial SIP provider -- would have to look like this: [jwinius] type=friend host=dynamic secret=passwrd insecure=invite Unfortunately, this doesn't work, resulting Failed to authenticate on INVITE errors. It only works if I first remove the fromuser and secret options from the configuration on the first system, but that's not what I want. Any idea what I'm doing wrong? Thanks, Jaap -- _ -- 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] Simulating a commercial SIP provider
If I understand well - you want second PBX to act as your sip.provider.com add this to your /etc/hosts (on primary pbx): 10.10.10.10 sip.provider.com (secondary - simulation pbx): 127.0.0.1 localhost sip.provider.com And use primary pbx as normal. When you need to switch to production - remove the /etc/hosts line. Hope this helps. On Mon, May 10, 2010 at 2:57 PM, Jaap Winius jwin...@umrk.to wrote: Hi all, The kind of configuration that I use in my sip.conf to connect to various commercial SIP providers looks like this: [general] context=incoming-calls canreinvite=no qualify=yes register = jwinius:pass...@sip.provider.com/0201234567 [provider] type=peer host=sip.provider.com fromuser=jwinius secret=passwrd This works. However, how would I have to configure the sip.conf of a second Asterisk machine if I wanted to use it to simulate the host mentioned above, sip.provider.com, but (crucially) without changing the above configuration? I would have thought that the appropriate stanza to use for my account in the other Asterisk machine's sip.conf -- the system that simulates the commercial SIP provider -- would have to look like this: [jwinius] type=friend host=dynamic secret=passwrd insecure=invite Unfortunately, this doesn't work, resulting Failed to authenticate on INVITE errors. It only works if I first remove the fromuser and secret options from the configuration on the first system, but that's not what I want. Any idea what I'm doing wrong? Thanks, Jaap -- _ -- 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] Simulating a commercial SIP provider
Forgot to mention that 10.10.10.10 must be the address of your secondary pbx :) 2010/5/10 Motiejus Jakštys desired@gmail.com If I understand well - you want second PBX to act as your sip.provider.com add this to your /etc/hosts (on primary pbx): 10.10.10.10 sip.provider.com (secondary - simulation pbx): 127.0.0.1 localhost sip.provider.com And use primary pbx as normal. When you need to switch to production - remove the /etc/hosts line. Hope this helps. On Mon, May 10, 2010 at 2:57 PM, Jaap Winius jwin...@umrk.to wrote: Hi all, The kind of configuration that I use in my sip.conf to connect to various commercial SIP providers looks like this: [general] context=incoming-calls canreinvite=no qualify=yes register = jwinius:pass...@sip.provider.com/0201234567 [provider] type=peer host=sip.provider.com fromuser=jwinius secret=passwrd This works. However, how would I have to configure the sip.conf of a second Asterisk machine if I wanted to use it to simulate the host mentioned above, sip.provider.com, but (crucially) without changing the above configuration? I would have thought that the appropriate stanza to use for my account in the other Asterisk machine's sip.conf -- the system that simulates the commercial SIP provider -- would have to look like this: [jwinius] type=friend host=dynamic secret=passwrd insecure=invite Unfortunately, this doesn't work, resulting Failed to authenticate on INVITE errors. It only works if I first remove the fromuser and secret options from the configuration on the first system, but that's not what I want. Any idea what I'm doing wrong? Thanks, Jaap -- _ -- 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
[asterisk-users] Records sets and ODBC
Hi, I have a system using ODBC and connecting to a MS-SQL database. Does anyone know if it is possible to return a record set consisting of several rows from SQL back into Asterisk? I have tried using ARRAY but only the contents of the last row are being stored. Thanks Lee -- _ -- 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] Unable to hear voice when called from PSTN phone line
Hi, My asterisk server is running behind a NAT/Firewall and I am using Skype for Asterisk. I am able to communicate using two SIP extension one connected from within the LAN and the other connected from outside my network. The problem is when I am trying to communicate using a PSTN phone I am able to establish a connection but I am not able to hear anything. Can someone help me to solve this issue. Regards Supratik -- _ -- 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] Continue dialplan is source channel hangs up
Hi, does anyone know if there is an equivalent dial option for the source channel to the g option? I've had a good look and can't find one. g- Proceed with dialplan execution at the current extension if the destination channel hangs up. Thanks Lee -- _ -- 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] Sometimes called party answers, but callee keep hear ringing, called party hears nothing!
Hi, As mentioned we have the problem that sometimes (could be up to a view times a day) for the calling party (SIP Device) you here ringing. The called party however answered the phone, but hears nothing. The calling party keeps ringing until the phone is hangup. First I thought maybe the card or the server has a problem, so I changed from a PCI beronet 4bri to a Junghanns 4bri PCIexpress and changed the server (as it was on a spare server anyway) from an old HP ProLiant DL360 G3 to an Dell PowerEdge 1850 with PCIe riser card. But the issue remains, it never occured with the old Siemens Hipath 3000! There are approx. 14 active phones and 6 analog devices connected to a Sipura SPA 3000. The Problem originates from the SIP Phones (can not tell about the analog devices, but they are, from asterisk standpoint, SIP as well). I experienced this behavior a few times as well. Log entries show nothing peculiar, but I also do not have Zap/Dahdi debugging active. The call get's passed to the DAHDI Device and there are no further indications of problems. Please tell me if more information is needed to fix this behavior, as this is in my experience not normal (other Asterisk solutions I administrate work fine). We are located in Spain, if this matters. we have an ubuntu server 32bit, stock ubuntu asterisk and dahdi: asterisk, 1:1.6.2.5-0ubuntu1 dahdi, 1:2.2.1-0ubuntu2 We have 4 ISDN lines, currently 3 lines connected. Some are connected as bri_cpe (from the former input to a Siemens HiPath 3000), and some as bri_cpe_ptmp. Span 2 currently not connected. Span configuration: ; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS group=0,11 context=from-zaptel switchtype = euroisdn signalling = bri_cpe channel = 1-2 context = default group = 63 ; Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 AMI/CCS RED group=0,12 context=from-zaptel switchtype = euroisdn signalling = bri_cpe_ptmp channel = 4-5 context = default group = 63 ; Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 AMI/CCS group=0,13 context=from-zaptel switchtype = euroisdn signalling = bri_cpe channel = 7-8 context = default group = 63 ; Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 AMI/CCS group=0,14 context=from-zaptel switchtype = euroisdn signalling = bri_cpe_ptmp channel = 10-11 context = default group = 63 ; Span 5: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: HRtimer) 1 Best, Ray -- _ -- 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] Getting presence working in 1.6.2
On Sunday 09 May 2010 23:20:15 Dovid Bender wrote: - Original Message - From: Tilghman Lesher tles...@digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 07, 2010 22:39 Subject: Re: [asterisk-users] Getting presence working in 1.6.2 On Friday 07 May 2010 13:59:23 Matt Darnell wrote: On Fri, May 7, 2010 at 3:41 AM, Jared Smith jsm...@digium.com wrote: To make it more clear and less cryptic, we split out the callcounter functionality in sip.conf, so that you could turn on/off the SIP device state tracking without limiting calls, and encouraged people to use the GROUP() and GROUP_COUNT() functions in the dialplan to enforce call limits. But why 'callcounter', it is frustratingly close 'call-limit' and there is no possible way to use logic to determine what it does. If a change was to be made, why not use 'devicestatetracking=yes'? As it's now in three releases, it's rather late to be changing it, although we could add an additional alias. You should probably be watching the commits list and send an email to the -dev list anytime you see something that you think could be better named. I find this very frustrating as well with Asterisk. I understand that names change for the better but I think there should be a discussion on it so all the users can come up with something that makes it easier for the most of us. There WAS a discussion of this commit on the -dev list, starting here: http://lists.digium.com/pipermail/asterisk-dev/2007-November/030790.html It's equally frustrating for developers to discuss this on public lists, then have people complain that there was no public discussion. We cannot force you to pay attention to discussions, can we? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Continue dialplan is source channel hangs up
F([[context^]exten^]priority): When the caller hangs up, transfer the called party to the specified destination and continue execution at that location. Also just F will continue to the next priority on the dialplan. De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de Lee Archer Enviado el: lunes, 10 de mayo de 2010 9:36 Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] Continue dialplan is source channel hangs up Hi, does anyone know if there is an equivalent dial option for the source channel to the g option? Ive had a good look and cant find one. g- Proceed with dialplan execution at the current extension if the destination channel hangs up. Thanks Lee Se certificó que el correo entrante no contiene virus. Comprobada por AVG - www.avg.es Versión: 9.0.819 / Base de datos de virus: 271.1.1/2864 - Fecha de la versión: 05/10/10 03:26:00 -- _ -- 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] Dialing a SIP Peer without using register strin
Hi, I am new to this list and this is first time i m posting here. please help me out currently I am working on dialing a sip peer on an asterisk server from 2nd asterisk server. scenario is like this. on my system i am using this peer in sip.conf. [abc] type=peer username=abc secret=mysecret host=192.168.0.20 context=default dtmfmode=rfc2833 ;restrictcid=no canreinvite=yes insecure=invite,port disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm nat=yes qualify=yes and using following register string register = abc:mysec...@192.168.0.20 abc%3amysec...@192.168.0.20 now problem is that when i use register string everything goes ok. but when i remove register string call doesn't go as expected. I would like to know if there is any feature that i can use to call sip peer and authenticate is in dial command or some feature in sip.conf i dont wanna use register string. please help. regards, Nasir Javaid -- _ -- 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] Continue dialplan is source channel hangs up
Thanks. Is there no 1.4 equivalent or is this a feature of 1.6 only? Lee From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Sent: 10 May 2010 14:45 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Continue dialplan is source channel hangs up F([[context^]exten^]priority): When the caller hangs up, transfer the called party to the specified destination and continue execution at that location. Also just F will continue to the next priority on the dialplan. De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de Lee Archer Enviado el: lunes, 10 de mayo de 2010 9:36 Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] Continue dialplan is source channel hangs up Hi, does anyone know if there is an equivalent dial option for the source channel to the g option? I've had a good look and can't find one. g- Proceed with dialplan execution at the current extension if the destination channel hangs up. Thanks Lee Se certificó que el correo entrante no contiene virus. Comprobada por AVG - www.avg.es Versión: 9.0.819 / Base de datos de virus: 271.1.1/2864 - Fecha de la versión: 05/10/10 03:26:00 -- _ -- 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] DAHDI not detecting hangup
I've got an old analog PBX and I'm trying to connect an FXO port on my asterisk server to one of the extensions on the old PBX. This should work as en extension on the old PBX should be providing dialtone, battery current and ring voltage. However, when the old PBX hangs up asterisk doesn't appear to be detecting the hangup (the DAHDI channel stays in use). Can anyone help? Thanks, Brian -- _ -- 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] More clarification on outbound sip channels.
Jim, and all:
Thanks for the response.
If I can repeat what you are saying: you don't have to define the multiple
lines in sip.conf?
For comparison, with my current esi setup, we have 10 outgoing lines. If one
line is busy, then the service just rolls to the next number. This is set up
with the phone service.
That doesn't have to done with outgoing sip lines? Does the dialstatus have to
be checked when a user dials out?
I understand the incoming lines - we will have a block of DID numbers, and I
can check those and send to appropriate extensions.
Thanks all for helping to clarify. I have gotten a couple of users who haven't
been able to call out, and wasn't sure if I wasn't rolling over the sip lines
properly.
Best,
Eddie Mikell
From: Jim Dickensondicken...@cfmc.com
Subject: Re: [asterisk-users] Multiple SIP lines.
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:eda8102c-b255-46e0-940d-1ef217566...@cfmc.com
Content-Type: text/plain; charset=us-ascii
I think it is typical to have some limited number of outbound channels to your
SIP provider. You send all calls, up to your limit, to the same place. The
phone numbers your provider gave you are used to route inbound calls to your
asterisk box. You will typically have some limited number of inbound channels.
All people could call the same number, again controlled by the number of
channels your provider allows. A reason to have multiple inbound (DID) numbers
is so you can route each number to a specific dialplan extension. You might
route one number to the CEO of the company and the other to a voice tree that
allows the caller to specify the person's extension they want to talk with.
-- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On
May 7, 2010, at 11:17 AM, Eddie Mikell wrote:
All:
Still experimenting with the asterisk server for the company.
My local phone company has given me two sip numbers to experiment with,
say 444-456-1234 444-456-5678
Calling in and out works, and I've spread a couple of the phones out
with some co-workers.
My question is this: Do I have to define multiple sip lines in either
the sip.conf or the extensions.conf?
Now when I dial out, I just use
exten = _9.,1,DIAL(SIP/${EXTEN:1...@xx.tracfone.net).
How does it know which sip channel to use?
Hope that is clear.
Thanks for all the help.
Eddie Mikell
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Re: [asterisk-users] DAHDI not detecting hangup
Hanguponpolarityswitch=yes in dahdi.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian C. Huffman Sent: Monday, May 10, 2010 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI not detecting hangup I've got an old analog PBX and I'm trying to connect an FXO port on my asterisk server to one of the extensions on the old PBX. This should work as en extension on the old PBX should be providing dialtone, battery current and ring voltage. However, when the old PBX hangs up asterisk doesn't appear to be detecting the hangup (the DAHDI channel stays in use). Can anyone help? Thanks, Brian -- _ -- 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] DAHDI not detecting hangup
On Mon, May 10, 2010 at 09:40:36AM -0500, Danny Nicholas wrote: Hanguponpolarityswitch=yes in dahdi.conf? chan_dahdi.conf, that is. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] DAHDI not detecting hangup
Can I apply that to only one channel? I'm on trixbox and I'm not sure which file should get these changes: /etc/dahdi/system.conf /etc/asterisk/chan_dahdi.conf /etc/asterisk/dahdi-channels.conf Thanks, Brian On Mon, 2010-05-10 at 09:40 -0500, Danny Nicholas wrote: Hanguponpolarityswitch=yes in dahdi.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian C. Huffman Sent: Monday, May 10, 2010 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI not detecting hangup I've got an old analog PBX and I'm trying to connect an FXO port on my asterisk server to one of the extensions on the old PBX. This should work as en extension on the old PBX should be providing dialtone, battery current and ring voltage. However, when the old PBX hangs up asterisk doesn't appear to be detecting the hangup (the DAHDI channel stays in use). Can anyone help? Thanks, Brian -- _ -- 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] More clarification on outbound sip channels.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 10, 2010, at 7:35 AM, Eddie Mikell wrote:
Jim, and all:
Thanks for the response.
If I can repeat what you are saying: you don't have to define the multiple
lines in sip.conf?
Your SIP provider will limit the number of concurrent outbound calls you can
make. If you try to dial more than allowed you will get a SIP message with some
error indicating all outbound channels in use.
For comparison, with my current esi setup, we have 10 outgoing lines. If one
line is busy, then the service just rolls to the next number. This is set up
with the phone service.
That doesn't have to done with outgoing sip lines? Does the dialstatus have
to be checked when a user dials out?
SIP calls set setup by talking to your SIP provider. They take care of limiting
concurrency. Both inbound and outbound. You can have logic in your dialplan
using functions GROUP and GROUP_COUNT to keep track of how many channels you
are using. Doing this allows you to play a sound file saying all lines are busy
try your call later.
If the dial command fails then ${DIALSTATUS} will have values like CHANUNAVAIL
CONGESTION NOANSWER BUSY ANSWER CANCEL DONTCALL TORTURE INVALIDARGS
I understand the incoming lines - we will have a block of DID numbers, and I
can check those and send to appropriate extensions.
Thanks all for helping to clarify. I have gotten a couple of users who
haven't been able to call out, and wasn't sure if I wasn't rolling over the
sip lines properly.
Best,
Eddie Mikell
From: Jim Dickensondicken...@cfmc.com
Subject: Re: [asterisk-users] Multiple SIP lines.
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:eda8102c-b255-46e0-940d-1ef217566...@cfmc.com
Content-Type: text/plain; charset=us-ascii
I think it is typical to have some limited number of outbound channels to
your SIP provider. You send all calls, up to your limit, to the same place.
The phone numbers your provider gave you are used to route inbound calls to
your asterisk box. You will typically have some limited number of inbound
channels. All people could call the same number, again controlled by the
number of channels your provider allows. A reason to have multiple inbound
(DID) numbers is so you can route each number to a specific dialplan
extension. You might route one number to the CEO of the company and the other
to a voice tree that allows the caller to specify the person's extension they
want to talk with.
-- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On
May 7, 2010, at 11:17 AM, Eddie Mikell wrote:
All:
Still experimenting with the asterisk server for the company.
My local phone company has given me two sip numbers to experiment with,
say 444-456-1234 444-456-5678
Calling in and out works, and I've spread a couple of the phones out
with some co-workers.
My question is this: Do I have to define multiple sip lines in either
the sip.conf or the extensions.conf?
Now when I dial out, I just use
exten = _9.,1,DIAL(SIP/${EXTEN:1...@xx.tracfone.net).
How does it know which sip channel to use?
Hope that is clear.
Thanks for all the help.
Eddie Mikell
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Re: [asterisk-users] DAHDI not detecting hangup
The file that should get the change is /etc/asterisk/chan_dahdi.conf. you can apply to one channel by doing this Hanguponpolarityswitch=yes Channel=1 Hanguponpolarityswitch=no Channel=2-x -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian C. Huffman Sent: Monday, May 10, 2010 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI not detecting hangup Can I apply that to only one channel? I'm on trixbox and I'm not sure which file should get these changes: /etc/dahdi/system.conf /etc/asterisk/chan_dahdi.conf /etc/asterisk/dahdi-channels.conf Thanks, Brian On Mon, 2010-05-10 at 09:40 -0500, Danny Nicholas wrote: Hanguponpolarityswitch=yes in dahdi.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian C. Huffman Sent: Monday, May 10, 2010 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI not detecting hangup I've got an old analog PBX and I'm trying to connect an FXO port on my asterisk server to one of the extensions on the old PBX. This should work as en extension on the old PBX should be providing dialtone, battery current and ring voltage. However, when the old PBX hangs up asterisk doesn't appear to be detecting the hangup (the DAHDI channel stays in use). Can anyone help? Thanks, Brian -- _ -- 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
[asterisk-users] shoretel pbx
I am using asterisk connected to shoretel pbx using SIP trunk that then is connected to PRI to the world. connection is there and I can make calls out. HOwever, as soon as I place the call with a call file it is telling me the call was answered and my cell phone has not even rang yet. What is not set right on the shortel end to provide me call progress on the PRI connection - not the SIP connection. I need to know when the call is answered on the PRI side not the SIP side. Thanks, Jerry -- _ -- 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] Records sets and ODBC
On Monday 10 May 2010 07:19:34 Lee Archer wrote: Hi, I have a system using ODBC and connecting to a MS-SQL database. Does anyone know if it is possible to return a record set consisting of several rows from SQL back into Asterisk? I have tried using ARRAY but only the contents of the last row are being stored. Only in 1.6.x (mode=multirow). -- Tilghman -- _ -- 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] Getting presence working in 1.6.2
At 09:20 PM 5/9/2010, you wrote: I find this very frustrating as well with Asterisk. I understand that names change for the better but I think there should be a discussion on it so all the users can come up with something that makes it easier for the most of us. Sorry to strongly disagree. The decision needs to be made by one person or a small group who see what Asterisk is and where it's going. Just because they got it wrong the first time and you've learned it does not make it right. And in my experience projects directed by the users without the oversight of a very small group with the projects best interest and a clear vision of the future become lost and unusable. I think the best projects come when the leaders are smart enough to say, We've heard what you want, here's what you need. Why no one stood up and said What are you thinking when the word core was added to the syntax escapes me as it seems to have been a giant step backwards. Adding the alias file seems like it makes the problem even worse, while I can now comfortably work on my Asterisk box, I will be completely lost on anyone else's box. Defining a language is a non-trivial problem and getting it wrong is ever so much easier than getting it right Ira -- _ -- 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] Manipulating the Blacklist database
I am running Asterisk 1.4.2 and recently we changed the SIP provider of our main incoming DID number. The new provider prefixes all CallerID records with a +1 in front of the number, whereas the previous SIP provider did not. Consequently now all my blacklisted numbers aren't matching in my Dialplan, so I'm getting tele-spammed. Is there a way that I can work with the blacklist database like a SQL database, and just apply a script to update all numbers and add the prefix to them if they don't have it already? Myles -- - Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA www.techsolusa.com Phone +1-480-451-7440 -- _ -- 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] Working with Blacklist database
I have an Asterisk 1.4.2 system online and have built up quite a large blacklist of tele-spammers that have been calling us. Recently we swapped one of our DID numbers to a SIP provider that now prefixes all calls with +1 in front of US numbers (we're in the USA) and I need to edit my blacklist database so that all numbers in there that don't have the +1 prefix in front of them, now have a version of it that can match to these new prefixes. Or alternatively I need to change my Dialplan to search for a match against the Blacklist database on a substring of the incoming caller ID. Has anyone done this before? Is it easy to edit the content of the Blacklist database like in a SQL database, where I can simply insert new records based on the old records but change the prefix to them with the +1? Myles -- - Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA www.techsolusa.com Phone +1-480-451-7440 -- _ -- 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] Pass MWI from analog line through DAHDI?
Is it possible to pass the Message Waiting indicator from an analog line to a DAHDI port and into asterisk (and to an extension)? Thanks, Brian -- _ -- 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] Working with Blacklist database
I have an Asterisk 1.4.2 system online and have built up quite a large
blacklist of tele-spammers that have been calling us. Recently we
swapped one of our DID numbers to a SIP provider that now prefixes all
calls with +1 in front of US numbers (we're in the USA) and
I need to edit my blacklist database so that all numbers in there that
don't have the +1 prefix in front of them, now have a version of it that
can match to these new prefixes.
Just remove the +1:
exten = xxx/_+1.,1,Set(CALLERID(num)=${CALLERID(num):2})
Where xxx is the relevant extension.
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[asterisk-users] Using Asterisk? Get on the press list!
I'll post my semi-annual request for press contacts. If you have an Asterisk installation that matches one or more of these adjectives: - enterprise-oriented - government-oriented - education-oriented - unique - clever - large - complex ...we would be interested in having you talk with the press! We at Digium keep a list of people who wouldn't mind talking to the press on their open-source Asterisk installations, because we get inquiries all the time about different things. We want to talk to people who are using Asterisk to do _! is what we get, and it's great to be able to have a pocket-full of people who meet the criteria who don't mind spending a bit of time talking about how they use Asterisk to solve a particular task. Remember: Asterisk and all open-source projects have almost zero press other than what YOU can help with. We're competing with tens (hundreds?) of millions of dollars in press and marketing that is spent by proprietary vendors. The press is very, very interested in hearing about Asterisk, but we don't have a giant list of OSS Asterisk users from which we can cherry-pick names, or offer discounts on yearly subscription fees (because there aren't any!) to help out with press interviews, etc. Help the project, help yourself - put your name on the list! If you've already filled out this form, no need to again - you're on the list, and if there is a press inquiry that seems to be a good match, we'll pass them to you! If you're an integrator, please don't fill out the form - have your customers fill out the form - press folks are always looking for end users, not integrators. http://bit.ly/asterisk-press JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.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] Simulating a commercial SIP provider
Quoting Motiejus Jak?tys desired@gmail.com: If I understand well - you want second PBX to act as your sip.provider.com add this to your /etc/hosts (on primary pbx): 10.10.10.10 sip.provider.com No, I'm afraid you misunderstand. This has nothing to do with DNS and not being able to reach my second PBX -- that's all fine. The hostname, sip.provider.com, is fictitious anyway. The problem is how to configure the client entry in the second PBX's sip.conf so that the first PBX can use it without having to change anything (other than the hostname). As things stand, I can already do it, but only if I first remove the fromuser and secret options from the sip.conf of the first PBX: that's going too far. Eventually, I hope to use the new information to expand this article: * Asterisk: minimal SIP configuration http://www.rjsystems.nl/en/2100-asterisk.php The text would start with: If a second Asterisk server is used to simulate the connection to the commercial SIP provider, add this stanza to its sip.conf ... Thanks anyway, Jaap -- _ -- 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] Manipulating the Blacklist database
On Mon, 10 May 2010, Myles Wakeham wrote:
I am running Asterisk 1.4.2 and recently we changed the SIP provider of
our main incoming DID number. The new provider prefixes all CallerID
records with a +1 in front of the number, whereas the previous SIP
provider did not.
Consequently now all my blacklisted numbers aren't matching in my
Dialplan, so I'm getting tele-spammed.
Probably all those left-leaning weenies that think AZ doesn't have the
right to remove criminals in the face of a disinterested federal
government :)
Is there a way that I can work with the blacklist database like a SQL
database, and just apply a script to update all numbers and add the
prefix to them if they don't have it already?
Myles
--
-
Myles Wakeham
Director of Engineering
Tech Solutions USA, Inc.
Scottsdale, Arizona USA
www.techsolusa.com
Phone +1-480-451-7440
Off the top of my head...
mysql create table foo as select * from blacklisted_anis;
mysql update foo set ani = concat('+1', ani) where ani not like '+1%';
mysql select * from foo limit 10;
If you like what you see, roll the dice with your production database.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] text
On Fri, 7 May 2010, Thomas Perron wrote: do i need to have an smtp server somewhere. i tried directly from my dialplan but no joy! i know you know that i am not a star with this but any help would be cool Start small and work your way up. From a shell command line, try echo test | mail -s test thomas.per...@gmail.com If that doesn't work and you don't get any useful clues from the command output, start digging where your syslogd logs messages. Next, from a shell command line, try echo test | mail -s test 551...@txt.att.net Note that this recipient is specific to this carrier. If that works, it should work in Asterisk assuming you don't have any path or permissions issues. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Using Asterisk? Get on the press list!
I just filled out this form and it needs some updates... 1. Field for website 2. City? I put City + Country because I am not in the USA 3. Other type of use: Building Automation 4. Size of Installation? For those of us who are integrators should we total all our installations or just list the largest? ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Mon, May 10, 2010 at 2:32 PM, John Todd jt...@digium.com wrote: I'll post my semi-annual request for press contacts. If you have an Asterisk installation that matches one or more of these adjectives: - enterprise-oriented - government-oriented - education-oriented - unique - clever - large - complex ...we would be interested in having you talk with the press! We at Digium keep a list of people who wouldn't mind talking to the press on their open-source Asterisk installations, because we get inquiries all the time about different things. We want to talk to people who are using Asterisk to do _! is what we get, and it's great to be able to have a pocket-full of people who meet the criteria who don't mind spending a bit of time talking about how they use Asterisk to solve a particular task. Remember: Asterisk and all open-source projects have almost zero press other than what YOU can help with. We're competing with tens (hundreds?) of millions of dollars in press and marketing that is spent by proprietary vendors. The press is very, very interested in hearing about Asterisk, but we don't have a giant list of OSS Asterisk users from which we can cherry-pick names, or offer discounts on yearly subscription fees (because there aren't any!) to help out with press interviews, etc. Help the project, help yourself - put your name on the list! If you've already filled out this form, no need to again - you're on the list, and if there is a press inquiry that seems to be a good match, we'll pass them to you! If you're an integrator, please don't fill out the form - have your customers fill out the form - press folks are always looking for end users, not integrators. http://bit.ly/asterisk-press JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recording with extension and agent in queue
Hi! I am recording with asterisk and so far so good. Now I need to use in the name of recording wich extension that takes the call and the agent in the queue that takes the call/ Is there a way to know what extension and the agent that take the call in a queue for recording??? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* _ -- _ -- 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] Speech/DTMF mix?
Which speed recognition products will also recognize DTMF? In other words, I want to say Please speak or dial the conference number. Does Vestec allow that? LumenVox? Any other way? -- _ -- 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] voipmonitor.org
Date: Mon, 10 May 2010 09:39:55 +0200 From: v...@lam.cz To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] voipmonitor.org On 8.5.2010 00:40, Jeff Brower wrote: Martin- checkout new open source voipmonitor.org SIP packet sniffer. I've developed it for my telco company and I've decided to share it. Testing and contributions are welcome! VoIPmonitor is open source live network packet sniffer which analyze SIP and RTP protocol. It can run as daemon or analyzes already captured pcap files. For each detected VoIP call voipmonitor calculates statistics about loss, burstiness, latency and predicts MOS (Meaning Opinion Score) according to ITU-T G.107 E-model. These statistics are saved to MySQL database and each call is saved as pcap dump. Web PHP application (it is not part of open source sniffer) filters data from database and graphs latency and loss distribution. Voipmonitor also detects improperly terminated calls when BYE or OK was not seen. To accuratly transform latency to loss packets, voipmonitor simulates fixed and adaptive jitterbuffer. How many channels can it handle simultaneously? I've not tested limits but capturing 15 voip calls takes 3-4% on Core2 2.40GHz. Complexity in worst case is O(N^2) where N is number of calls. Packets are matched as llinear list of IP and port. If this will be limit, it could be rewriten to hash table O(N) How does it do MOS prediction if low bitrate codecs are being used (G729, AMR, etc)? It is calibrated only to G.711 with PLC for now but I'm planing adding equations for G.729 and iLBC. MV Maybe this question is out little but is the same context. I need read the VoIP packets and order all this packets in another place to get the audio. The idea is can record a call using directly the packets. I know asterisk can record but my problem is that I have Avaya and asterisk working togheter and I can not record by Avaya and somebody tells me this idea to sniff the VoIP packets order after the call. I am seeing the code for VoIp monitor Is it so stupid?? TIA _ -- _ -- 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] voipmonitor.org
On 05/10/2010 07:01 PM, Edwin Quijada wrote: Maybe this question is out little but is the same context. I need read the VoIP packets and order all this packets in another place to get the audio. The idea is can record a call using directly the packets. I know asterisk can record but my problem is that I have Avaya and asterisk working togheter and I can not record by Avaya and somebody tells me this idea to sniff the VoIP packets order after the call. I am seeing the code for VoIp monitor Is it so stupid?? There are already products available (both open source and commercial) that do this; look for OrecX. -- 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
Re: [asterisk-users] Speech/DTMF mix?
On Mon, May 10, 2010 at 7:19 PM, Richard Kenner ken...@gnat.com wrote: Which speed recognition products will also recognize DTMF? In other words, I want to say Please speak or dial the conference number. Does Vestec allow that? LumenVox? Any other way? You're on your own for making custom messaging. For everything else, there's: http://www.voip-info.org/wiki/view/Asterisk+cmd+Read http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigits -- _ -- 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] Speech/DTMF mix?
On Mon, May 10, 2010 at 7:19 PM, Richard Kenner ken...@gnat.com wrote: Which speed recognition products will also recognize DTMF? In other words, I want to say Please speak or dial the conference number. Does Vestec allow that? LumenVox? Any other way? You're on your own for making custom messaging. For everything else, there's: http://www.voip-info.org/wiki/view/Asterisk+cmd+Read http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigits Sorry, I had a typo: I meant speech, not speed. In other words, it is possible to use any speech recognition product to detect EITHER speech or DTMF? -- _ -- 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] AsteriskNow
H Group- I loaded the image via ISO and it loads fine to a command prompt (Linux prompt). I would like to know how to open a GUI interface from which to operate Asterisk. Is it possible? Dave Mynatt // http://sciencecenter.no-ip.org http://sciencecenter.no-ip.org/ SID ID #S-0258 APRS Tier II Server @ http://pueblo.aprs2.net:14501 http://pueblo.aprs2.net:14501/ 10 meter CW Beacon @ 28.2345 Mhz RMS Server: 144.950 Mhz // KA0SWT-10 Echolink Server Node: 473082 @147.480Mhz Simplex KA0SWT-L LAT: (38.15.49) 38.26349N LON: (104.36.48) -104.613297W Alt: 1460m DM78qg // KA0SWT // -- _ -- 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] Re TrixBox
Trixbox ce callcenter works ok but call management is worthless. HUD and the control panel are not made for call centers. If they ever get hud 3 working for ce it is suppose to have more call center stuff. Your going to have to look at 3rd party apps And yes trixbox questions need to be asked in the trixbox forums www.trixbox.org. Respectfully Michael D Mosier Ftoc Certified On May 9, 2010 8:24 AM, Samantha saman...@femtech.com.au wrote: Hey Guys We are replacing a BM4 with a trixbox (asterisk) virtual numbers as the customer wants to move the callcentre. They are asking for an equiv to the ipview I gather HUD may be or the panel view The problem is that we need to see (a) total calls in the queue (b) calls for specific DID - How can you give 1 DID preference to another DID ie DID 61740410001 = Fred Electrical DID 61740410002 = Bus Tour ABC DID 61740410003 = Fred 24/7 Plumber SO in A we need to see how many calls waiting So in B we need to see how many calls are waiting for DID 001 002 and 003 finally if there are 20 calls on hold in DID 001 and 002 and there is a call on 003, how can we place that to the top of the queue? thanks Samantha -- _ -- 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] AsteriskNow
This is probably better asked in the asterisknow list. Log into the console Type ifconfig to get ip Go to web browser and type in the ip like Http://xxx.xxx.xxx.xxx/admin I forgot the admin passwor google around for it. Also please read more befor posting there are plenty of sites that eould answer this its very basic Respectfully Michael D Mosier Ftoc Certified On May 10, 2010 9:37 PM, Dave d...@mynatt.biz wrote: H Group- I loaded the image via ISO and it loads fine to a command prompt (Linux prompt). I would like to know how to open a GUI interface from which to operate Asterisk. Is it possible? Dave Mynatt // http://sciencecenter.no-ip.org SID ID #S-0258 APRS Tier II Server @ http://pueblo.aprs2.net:14501 10 meter CW Beacon @ 28.2345 Mhz RMS Server: 144.950 Mhz // KA0SWT-10 Echolink Server Node: 473082 @147.480Mhz Simplex KA0SWT-L LAT: (38.15.49) 38.26349N LON: (104.36.48) -104.613297W Alt: 1460m DM78qg // KA0SWT // -- _ -- 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] Speech/DTMF mix?
I think Asterisk will detect the dtmf for you and the speach recognition will detect speach. Respectfully Michael D Mosier Ftoc Certified On May 10, 2010 9:24 PM, Richard Kenner ken...@gnat.com wrote: On Mon, May 10, 2010 at 7:19 PM, Richard Kenner ken...@gnat.com wrote: Which speed recogniti... Sorry, I had a typo: I meant speech, not speed. In other words, it is possible to use any speech recognition product to detect EITHER speech or DTMF? -- _ -- Bandwidth and Colocati... -- _ -- 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] Problem of Playing 'pbx-transfer'
Our codec is ulaw. We tested snom to snom, x-lite to x-lite. We are getting same problems as usual. I alse tested for another device like linksys to snom, x-lite to snom. mobile phone to snom or x-lite. The same problem occurs for above two options. We sometimes hear transfer' s sound, sometime cannot hear transfer sound. The same problem for answering machine. Is it the load problem or something else? Is there any solution for this problem? our system is like as: The number of User agent is: 1650 The number of Actual registered user agent is: 600 version of asterisk: 1.4.23.1 our memory size is 4GB. concurrent calls no : 30. Thanks in advance Nahar On Mon, May 10, 2010 at 8:25 PM, Adolphe Cher-aime achera...@gmail.comwrote: Try x-lite to x-lite, snom to snom . That may be a codec problem. Which codec are you using? Adolphe Cher-aime From my Iphone On May 9, 2010, at 11:11 PM, Dovid Bender asteriskus...@dovid.net wrote: Process of elemination. Test with multiple phones, check the codec being used and make sure the file is there and available. - Original Message - *From:* kamrun nahar bina bina...@gmail.com *To:* asterisk-users@lists.digium.comasterisk-users@lists.digium.com *Sent:* Friday, May 07, 2010 07:33 *Subject:* [asterisk-users] Problem of Playing 'pbx-transfer' Dear all, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer. During attended transfer, sometimes we cannot hear the sound of 'pbx-transfer'. I cannot understand why this is happening? log is : -- Started music on hold, class 'default', on SIP/113.34.235.13-b7a3f110 -- SIP/185148-092db338 Playing 'pbx-transfer' (language 'jp') Although it is showing Playing 'pbx-transfer' (language 'jp'), but it cannot hear 'pbx-transfer' sound Sometimes we can hear the sound of 'pbx-transfer'. is it the problem of network load or phone-set or something else? Please let me know. I am using x-lite and snom 300. Before i tested it for memory load, And found out that it is not a memory problem. Our system is as like as: The number of User agent is: 1650 The number of Actual registered user agent is: 600 Our System configuration is : IBM X3550 CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz HDD: 3.5 SATA 1TB x 2 version of asterisk: 1.4.23.1 our memory size is 4GB. concurrent calls no : 30. Our memory condition is below : Cpu(s): 0.3%us, 0.7%sy, 0.0%ni, 98.5%id, 0.0%wa, 0.1%hi, 0.3%si, 0.0%st Mem: 4147888k total, 3986540k used, 161348k free,76852k buffers Swap: 2031608k total, 56k used, 2031552k free, 3170396k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 23160 root 15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asterisk Our disk space condition is below: FilesystemSize Used Avail Use% Mounted on /dev/mapper/VolGroup00-LogVol00 901G 245G 610G 29% / /dev/sda1 99M 18M 77M 19% /boot tmpfs 2.0G 0 2.0G 0% /dev/shm Asterisk and the User-Agent is connected through the Internet. ..And Is there any solution to solve this problem? I have investigated in several places but I cannot find out the reason? I need this solution very urgently. Is there any one who can solve this problem? -- _ -- 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 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech/DTMF mix?
I think Asterisk will detect the dtmf for you and the speach recognition will detect speach. That's what I was hoping could be done. How do you set up the dialplan to have both of those functions run simultaneously? -- _ -- 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] AGI and Severe Weather Alerts
All, I am toying with an idea of using an AGI to be able to 'call' my phone, or phones, in case of severe weather warnings. I have been tinkering with a script that reads from weather underground for the forecast, based off a PHP version of a weather AGI I found on the net. It seems rather trivial to have the AGI as a script, that does nothing unless a condition is met, and then call out, with a TTS synthesized read out of the warning, or error seen. I would like to know if anyone has done this before and what they used to get the warning for their area's. I haven't a very clear idea of how to parse properly XML data in either python or perl, but I have templates of what did work (until formats changed, StormSiren being a python module I used for sms). Also if I ever get anything to work, and anyone is interested I can share my code. Regards, Seann Clark smime.p7s Description: S/MIME Cryptographic 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] Dialing a SIP Peer without using register strin
Remove username and secret and use IP authentication on both side [server1_abc] type=peer host=192.168.0.20 context=default dtmfmode=rfc2833 canreinvite=yes - canreinvite with nat=yes is not working insecure=invite,port disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm nat=yes qualify=yes [server2_abc] type=peer host=192.168.0.21 context=default dtmfmode=rfc2833 canreinvite=yes insecure=invite,port disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm nat=yes qualify=yes Nasir Javaid wrote: Hi, I am new to this list and this is first time i m posting here. please help me out currently I am working on dialing a sip peer on an asterisk server from 2nd asterisk server. scenario is like this. on my system i am using this peer in sip.conf. [abc] type=peer username=abc secret=mysecret host=192.168.0.20 context=default dtmfmode=rfc2833 ;restrictcid=no canreinvite=yes insecure=invite,port disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm nat=yes qualify=yes and using following register string register = abc:mysec...@192.168.0.20 mailto:abc%3amysec...@192.168.0.20 now problem is that when i use register string everything goes ok. but when i remove register string call doesn't go as expected. I would like to know if there is any feature that i can use to call sip peer and authenticate is in dial command or some feature in sip.conf i dont wanna use register string. please help. regards, Nasir Javaid -- _ -- 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
