[asterisk-users] Problems w/Asterisk Realtime + MySQL + SIP
Hi all, Asterisk is great but I'm having issues with setting up realtime for our call center, which is needed for login integration with the rest of our applications (telephonists' web interface, etc.). I have reviewed a large number of previous posts to the mailing list and the voip-info wiki to no avail. Setup is as follows: Linux 2.6.23 (gentoo) / AMD Athlon(tm) 64 Processor 3000+ / 2GB RAM Sangoma A102 card + E1 (30 channels) Asterisk 1.4.17 (custom compile from source, not using gentoo package or any patches) Wanpipe drivers 3.2.3.0 Iaxmodem + libiax 2-0.2.3-SVN-20071223+ I have tried both kphone and zoiper (linux) clients. On kphone the interface's register result is 'bad password', on zoiper registration continues indefinitely but after the first request it is ignored by asterisk due to being duplicate, after a time it fails silently. The debug log: [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: = No match Their Call ID: NDcxYjAyNTc4ZDQwZjZhMzM5OGE0MWYxYjg0YzZhZDk. Their Tag a9a71835 Our tag: as0a26e7a5 [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: Allocating new SIP dialog for ZjFhZjZlNmZmZjM3OWFlYzE0MGYzZDYwYzJmODAwNDg. - REGISTER (No RTP) [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: Received REGISTER (2) - Command in SIP REGISTER [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: = Found Their Call ID: ZjFhZjZlNmZmZjM3OWFlYzE0MGYzZDYwYzJmODAwNDg. Their Tag f1b0df07 Our tag: as25a61774 [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: Received REGISTER (2) - Command in SIP REGISTER [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: SIP message could not be handled, bad request: ZjFhZjZlNmZmZjM3OWFlYzE0MGYzZDYwYzJmODAwNDg. Console output: *CLI [Jul 21 15:40:47] DEBUG[2105]: chan_sip.c:4562 find_call: = Found Their Call ID: [EMAIL PROTECTED] Their Tag Our tag: as60d9fbbb [Jul 21 15:40:47] DEBUG[2105]: chan_sip.c:15154 handle_request: Received REGISTER (2) - Command in SIP REGISTER [Jul 21 15:40:47] NOTICE[2105]: chan_sip.c:15049 handle_request_register: Registration from 'walter sip:[EMAIL PROTECTED]' failed for '192.168.0.25' - Wrong password [Jul 21 15:40:47] DEBUG[2105]: chan_sip.c:15372 sipsock_read: SIP message could not be handled, bad request: [EMAIL PROTECTED] This error is different to the error that is received if a username that is not in the MySQL sip_peers / sip_users table is specified. Therefore at least the MySQL connection appears to be working. extconfig.conf: sipusers = mysql,asterisk_config sippeers = mysql,asterisk_config I have also tried explicitly adding ',sip_users' and ',sip_peers' to these lines, but asterisk behaved similarly. res_mysql.conf dbhost = 127.0.0.1 dbname = asterisk_config dbuser = asterisk dbpass =snip ;dbport = 3306 dbsock = /tmp/mysql.sock MySQL tables follow. They are static right now for debugging purposes, actually we will use views. We will use md5 passwords, but I have both in there right now for testing. mysql select * from sip_peers; ++++-+--+-+--+--++ | user | type | secret | host| context | pickupgroup | md5secret| username | name | ++++-+--+-+--+--++ | walter | friend | aaa| dynamic | outgoing | 1 | 47bce5c74f589f4867dbd57e9ca9f808 | walter | walter | ++++-+--+-+--+--++ 1 row in set (0.00 sec) mysql select * from sip_users; ++++-+--+-+--++--+ | user | type | secret | host| context | pickupgroup | md5secret| name | username | ++++-+--+-+--++--+ | walter | friend | aaa| dynamic | outgoing | 1 | 47bce5c74f589f4867dbd57e9ca9f808 | walter | walter | ++++-+--+-+--++--+ 1 row in set (0.00 sec) Thanks for any help you can offer. Regards, Walter Stanish Owner / Director Occident Systems (+86 15808 700 801) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OSLEC vs HPEC vs Octasic
Hello, I am trying to figure out which soft echo canceller I could use. There is OSLEC, HPEC from Digium and Octware from Octasic. I have problems to find details about their CPU needs. Can anyone share his experience. What CPU and Memory is required for 2,4,8 and 16 channels? Any help is appreciated. Best regards, Loic Didelot. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First-time queue app: verifying human member?
Matt Riddell wrote:
Erik Anderson wrote:
Good evening all - for the first time, I'm implementing my first-ever
queue in asterisk. Overall, it's a pretty simple setup, 4 static
members, very low call volume, etc. The one thing that has stumped me
so far, though, is the following...
This is a queue I'm setting up for contacting our IT support staff
off-hours. As such, I've just added the cell phone numbers of our
staff as members. I'd like to somehow verify that it's an actual human
answering the phone when a member is dialed and not their mobile
phone's voicemail. Is that possible? I'd envision just requesting that
the member press 1 or something to accept the call. I currently have
the timeout in queues.conf set low enough so that the call will never
automatically roll over to that member's mobile voicemail, but I can't
guaranty that the staff member won't just hit Ignore on their phone
and send it directly to voicemail.
You'd probably want to look at using the local channel and the followme
application + /etc/asterisk/followme.conf
Full details:
1) create an entry per engineer in followme.conf
2) add each engineer to your queue as Local/[EMAIL PROTECTED]
3) create a followme context in extensions.conf
=followme.conf=
[bob]
number=07973000123
[jim]
number=07973000124
=queue.conf=
[support]
member = Local/[EMAIL PROTECTED]
member = Local/[EMAIL PROTECTED]
=extensons.conf=
[meetme]
exten = ._,1,MeetMe(${EXTEN})
--
Will Tatam
***
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[asterisk-users] Problems with IAX on heartbeat provided ip address
Hi! I'm trying to build an HA system using heartbeat for failover. Everything works fine with SIP, but I cannot connect my IAX phone to the asterisk server using the managed IP address. Here is the configuration of the server (asterisk and the IP address are up, 'ip addr' and 'netstat' output): 2: eth0: BROADCAST,MULTICAST,UP,LOWER_UP mtu 1500 qdisc pfifo_fast qlen 1000 link/ether 52:54:00:51:3b:e2 brd ff:ff:ff:ff:ff:ff inet 10.241.85.80/24 brd 10.241.85.255 scope global eth0 inet 10.241.85.201/24 brd 10.241.85.255 scope global secondary eth0:0 inet6 fe80::5054:ff:fe51:3be2/64 scope link valid_lft forever preferred_lft forever tcp0 0 0.0.0.0:50380.0.0.0:* LISTEN 28144/asterisk tcp0 0 0.0.0.0:20000.0.0.0:* LISTEN 28144/asterisk udp0 0 0.0.0.0:27270.0.0.0:* 28144/asterisk udp0 0 0.0.0.0:45200.0.0.0:* 28144/asterisk udp0 0 0.0.0.0:50600.0.0.0:* 28144/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 28144/asterisk Attaching wireshark shows that the IAX phone never receives any response from the asterisk server to its 'REGREQ'. The IAX connection works fine as soon as I connect to '10.241.85.80' instead of '10.241.85.201'. There are no firewall rules in place (iptables is not even installed). Any ideas? Does someone know of any relevant bugs in asterisk 1.4.17? Cheers, Florian -- DI Florian Hackenberger [EMAIL PROTECTED] www.hackenberger.at ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC vs HPEC vs Octasic
2008/7/21 Loic Didelot [EMAIL PROTECTED]: Hello, I am trying to figure out which soft echo canceller I could use. There is OSLEC, HPEC from Digium and Octware from Octasic. I thought HPEC was licenced by Digium from Octasic (ie those 2 software are the same). Maybe someone should correct me ... I have problems to find details about their CPU needs. Can anyone share his experience. What CPU and Memory is required for 2,4,8 and 16 channels? Any help is appreciated. Best regards, Loic Didelot. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC vs HPEC vs Octasic
Olivier wrote: I thought HPEC was licenced by Digium from Octasic (ie those 2 software are the same). Maybe someone should correct me ... That is not correct; HPEC is a G.168 line echo canceller from Adaptive Digital Technologies. The same algorithm (but not the same source code) is used on the VPMADT032 module, which is available for all Digium analog line interface cards and single-span T1/E1/J1 interface cards. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC vs HPEC vs Octasic
On Mon, 21 Jul 2008, Loic Didelot wrote: Hello, I am trying to figure out which soft echo canceller I could use. There is OSLEC, HPEC from Digium and Octware from Octasic. I have problems to find details about their CPU needs. Can anyone share his experience. What CPU and Memory is required for 2,4,8 and 16 channels? Any help is appreciated. I switched to OSLEC after testing HPEC on TDM400 boards, and found that it worked much better and wasn't limited to the restricted mechanism Digium uses for licensing (unlikely as it sounds, I have some clients who do not have a connection to the public Internet, and never will for their phone system) It also passes the wife test which HPEC didn't. It's also free (OS as in Open Source), which HPEC isn't, although that wasn't my primary reason for using it - ease of use and workability was. As far as CPU usage is concerned, OSLEC gave me the tools to find that out - I didn't find any such tools with HPEC, but they might be there somewhere. On one of my production PBXs - a 1GHz VIA processor, 128KB cache, OSLEC can do the following: (running their own speedtest program) Testing OSLEC with 128 taps (16 ms tail) CPU executes 996.06 MIPS - Method 1: gettimeofday() at start and end 268 ms for 10s of speech 26.69 MIPS 37.31 instances possible at 100% CPU load Method 2: samples clock cycles at start and end 26.69 MIPS 37.31 instances possible at 100% CPU load Method 3: samples clock cycles for each call, IIR average cycles_worst 186709 cycles_last 43447 cycles_av: 4272 34.18 MIPS 29.15 instances possible at 100% CPU load So at worst, it's saying it can handle 29 incarnations, and at best, 37 - that's assuming no other CPU load such as transcoding. So it's well capable of handing your requirements of 16 channels - more-so if you're using a server class box, and not the embedded type systems I'm using here. (On my dev box, an older 2GHz Celeron, 128KB cache, it's telling me it can do 120 incarnations, and on a 2.4GHz Xeon with 4MB cache, it said it could do 321) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems w/Asterisk Realtime + MySQL + SIP
On Mon, Jul 21, 2008 at 9:11 AM, Walter Stanish [EMAIL PROTECTED] wrote: [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: Received REGISTER (2) - Command in SIP REGISTER [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: SIP message could not be handled, bad request: ZjFhZjZlNmZmZjM3OWFlYzE0MGYzZDYwYzJmODAwNDg. Hi Walter. It looks like Asterisk is unhappy with the SIP REGISTER request coming from your softphone for some reason. It's very strange that it's occurring for two different softphones though. Trun on SIP debugging by typing sip debug on your Asterisk console and then post up the 4 SIP messages invloved in the register transaction so we can take a look and spot why it could be getting rejected. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC vs HPEC vs Octasic
Gordon Henderson wrote: So at worst, it's saying it can handle 29 incarnations, and at best, 37 - that's assuming no other CPU load such as transcoding. So it's well capable of handing your requirements of 16 channels - more-so if you're using a server class box, and not the embedded type systems I'm using here. (On my dev box, an older 2GHz Celeron, 128KB cache, it's telling me it can do 120 incarnations, and on a 2.4GHz Xeon with 4MB cache, it said it could do 321) Those numbers are with a 16ms tail, which is very short, and unlikely to be an adequate echo tail for connection to the PSTN (although fine for analog phones). A more normal configuration would be 32, 64 or 128 millisecond tails, which would cut those numbers down by a factor of 2, 4 or 8. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC vs HPEC vs Octasic
Kevin P. Fleming wrote: Olivier wrote: I thought HPEC was licenced by Digium from Octasic (ie those 2 software are the same). Maybe someone should correct me ... That is not correct; HPEC is a G.168 line echo canceller from Adaptive Digital Technologies. The same algorithm (but not the same source code) is used on the VPMADT032 module, which is available for all Digium analog line interface cards and single-span T1/E1/J1 interface cards. G.168 is not an algorithm. Its a test spec. These cancellers all use related, but different, algorithms. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC vs HPEC vs Octasic
Thank you for you answers. So what tail would be suggested for SIP - LOCAL LAN - Asterisk - ISDN/BRI ? Is HPEC more or less resource hungry compared to OSLEC? Best regards, Loic Didelot. On Mon, 2008-07-21 at 06:54 -0500, Kevin P. Fleming wrote: Gordon Henderson wrote: So at worst, it's saying it can handle 29 incarnations, and at best, 37 - that's assuming no other CPU load such as transcoding. So it's well capable of handing your requirements of 16 channels - more-so if you're using a server class box, and not the embedded type systems I'm using here. (On my dev box, an older 2GHz Celeron, 128KB cache, it's telling me it can do 120 incarnations, and on a 2.4GHz Xeon with 4MB cache, it said it could do 321) Those numbers are with a 16ms tail, which is very short, and unlikely to be an adequate echo tail for connection to the PSTN (although fine for analog phones). A more normal configuration would be 32, 64 or 128 millisecond tails, which would cut those numbers down by a factor of 2, 4 or 8. -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue members randomly become paused after upgrade to Asterisk 1.4
Hi all, I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems that sometimes some phones become paused and cannot receive calls anymore. I tried to set autopause = no in every section of my queues.conf but nothing changes Anybody knows why a phone becomes paused? Is it an Asterisk 1.4 bug or there is a particular reason for this behaviour? Thank you. Giorgio. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC vs HPEC vs Octasic
Loic Didelot wrote: Thank you for you answers. So what tail would be suggested for 64ms Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC vs HPEC vs Octasic
Kevin P. Fleming wrote: Gordon Henderson wrote: So at worst, it's saying it can handle 29 incarnations, and at best, 37 - that's assuming no other CPU load such as transcoding. So it's well capable of handing your requirements of 16 channels - more-so if you're using a server class box, and not the embedded type systems I'm using here. (On my dev box, an older 2GHz Celeron, 128KB cache, it's telling me it can do 120 incarnations, and on a 2.4GHz Xeon with 4MB cache, it said it could do 321) Those numbers are with a 16ms tail, which is very short, and unlikely to be an adequate echo tail for connection to the PSTN (although fine for analog phones). A more normal configuration would be 32, 64 or 128 millisecond tails, which would cut those numbers down by a factor of 2, 4 or 8. If you try capturing echoes from real phone calls you will find very few exceeding 16ms, even for long distance calls. This is probably because the network has a canceller which you can't normally disable for a voice call. If you send a 2100Hz beep at the start of the call, you may then see much longer echoes appear. This is the echo canceller disable tone, which modems send to clear any networks cancellers from the line. The nature of modems means they need to do their own end to end cancellation, and that canceller certainly does need to cover a lot more than 16ms. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel and callerid in ESTI DTMF
Hello! I'm using Asterisk 1.4.18 (I've tried 1.4.19,1.4.21 too) and zaptel version 1.4.11. Card is Digium Wildcard TDM800P, with driver wctdm24xxp. From Asterisk side this card has FXS ports, and FXO from outside. I've connect to them GSM-FXO gateway Benq C5 APC-868 (http://www.kontec.ru/c5.php). The problem is that this equipment sends caller id information in format 'ESTI DTMF', which is not compatible with parameter 'cidsignalling = dtmf' in 'zapata.conf'. So, Asterisk didn't receive callerid information from device. I've tried to use all available zones in zaptel.conf and all allowed parametrs in 'cidsignalling' and 'cidstart' but there were no differences. Are anyone has faced with this or similar trouble? How to force Asterisk (may be by editing/tuning the sources)? -- Denis V. Gudtsov JSC Tango TELECOM tel +7 (3412) 916-500, 916-503 icq# 158668135 [EMAIL PROTECTED] ; www.tangotel.ru ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue() AGI Bug ?
Will Tatam wrote: The docs state that the AGI is run when the caller is connected but this does not appear to be true with 1.4.21.1 What I see is 1) caller enters queue 2) agent is found for call 3) agent1's call begins to ring 4) AGI is executed 5) agent does not answer the call before timeout, call goes to next agent 6) agent2 answers call but the AGI has already run Expected behaviour 1) caller enters queue 2) agent is found for call 3) agent1's call begins to ring 4) agent does not answer the call before timeout, call goes to next agent 5) agent2 answers call but the AGI has already run 6) AGI is executed I need the AGI to run when the actual call is connected to an agent as my AGI is tracking which agent took the call to then fire of a jabber message to that agent giving them them the url to access the caller's account page. Currently the message is going to agent1 and agent2 who actually takes the call never sees the message What type of channels do you use for your agents? If you're using Agent channels (the type which are configured in agents.conf), are you logging them in using AgentCallbackLogin? Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue members randomly become paused after upgrade to Asterisk 1.4
Giorgio Incantalupo wrote: Hi all, I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems that sometimes some phones become paused and cannot receive calls anymore. I tried to set autopause = no in every section of my queues.conf but nothing changes Anybody knows why a phone becomes paused? Is it an Asterisk 1.4 bug or there is a particular reason for this behaviour? Thank you. Giorgio. Are you sure that the phones in question are actually paused? What is displayed when running the queues show command from the CLI? It could be that the device state for the queue member has become stuck. What types of channels do you use for your queue members? Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC vs HPEC vs Octasic
Steve Underwood wrote: G.168 is not an algorithm. Its a test spec. These cancellers all use related, but different, algorithms. Yeah, that's what I get for emailing before breakfast :-) There is a missing 'compliant' in that sentence... -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incompatible voice frame panic!
Hi all, Panic! Panic! When I get a call over mISDN to my IAX extension and try to transfer it to another IAX/SIP, I get this message: Dropping incompatible voice frame on ... of format ulaw since our native format has changed to alaw Immediately followed by one almost the same: Dropping incompatible voice frame on ... of format alaw since our native format has changed to ulaw and so on, and so forth... Any ideas??? Thanks, David Vazquez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with IAX on heartbeat provided ip address
Florian Hackenberger wrote: Hi! I'm trying to build an HA system using heartbeat for failover. Everything works fine with SIP, but I cannot connect my IAX phone to the asterisk server using the managed IP address. I've had a similar issue with HA, although in my case SIP wouldn't register either. In my case, it was fixed by including one bindaddr=x.x.x.x statement in the [general] section of iax.ocnf per IP address that the machine could respond on. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Option 't' on DIal
Hi,
I encountered something i can't understand. I've setup 2 extensions.
[100]
type=friend
host=dynamic
nat=yes
secret=100
[101]
type=friend
host=dynamic
nat=yes
secret=101
and on extensions.conf
exten = _1XX,1,Dial(SIP/${EXTEN}|30|t)
exten = _1XX,n,Hangup
This dial plan is ok, audio connects both ways.
but when i had a typo error, i forgot the 't' option, only one way audio
when i call, 't' option is used to transfer call how come it affected
the audio?
thank you in advanced
regards
nhadie
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[asterisk-users] Recommend quality wholesale termination - Singapore and Sydney, Aus
Can anyone recommend decent quality as close to pay-as-you-go SIP wholesale termination providers in both Singapore and Sydney, Australia? I will be in both places and want a local carrier while I'm there. It needs to be easy in and easy out and if it's not $0 base or close I'll need to be able to drop it in a month. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue members randomly become paused after upgrade to Asterisk 1.4
Hi Mark, it is show queues I use to see if phones are paused or not. The phones I'm using for tests are all SIP phones. Yes, what you are supposing could be right...Asterisk could see the phones as stuck. I'm still investigating, making test on my 1.4 box and I have noticed some other strange things about the phones. Some phones when normally used (I made a test making an outbound call) are seen as paused (In use) while other are marked as In Use only: (from Asterisk CLI): SIP/8 with penalty 1 (In use) has taken 1 calls (last was 3247 secs ago)(my phone) SIP/36 with penalty 1 (paused) (In use) has taken no calls yet(my test phone) The phones are the same model and have same sip.conf definition. The queues.conf definitions are the same for the two queues the phones are in. I do not know why queues show shows paused or not for similar phones. Can this be useful!?!? Giorgio Mark Michelson wrote: Giorgio Incantalupo wrote: Hi all, I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems that sometimes some phones become paused and cannot receive calls anymore. I tried to set autopause = no in every section of my queues.conf but nothing changes Anybody knows why a phone becomes paused? Is it an Asterisk 1.4 bug or there is a particular reason for this behaviour? Thank you. Giorgio. Are you sure that the phones in question are actually paused? What is displayed when running the queues show command from the CLI? It could be that the device state for the queue member has become stuck. What types of channels do you use for your queue members? Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Option 't' on DIal
Nhadie wrote:
Hi,
I encountered something i can't understand. I've setup 2 extensions.
[100]
type=friend
host=dynamic
nat=yes
secret=100
[101]
type=friend
host=dynamic
nat=yes
secret=101
and on extensions.conf
exten = _1XX,1,Dial(SIP/${EXTEN}|30|t)
exten = _1XX,n,Hangup
This dial plan is ok, audio connects both ways.
but when i had a typo error, i forgot the 't' option, only one way audio
when i call, 't' option is used to transfer call how come it affected
the audio?
thank you in advanced
regards
nhadie
The 't' option is one that requires Asterisk to be in the media path of the
call
(so that Asterisk can tell when the transfer DTMF has been pressed). In order
to
stay in the path, SIP reinvites are disabled for the call. Without the 't'
option, Asterisk will send reinvites to the phones so that their media does not
go through Asterisk at all.
In order to figure out why there is one-way audio, you would need to provide a
sip debug of the call. Based on the fact that you have nat=yes for both SIP
friends, I'm guessing that there's some sort of NAT issue here, but I can't be
certain.
Mark Michelson
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Re: [asterisk-users] queue members randomly become paused after upgrade to Asterisk 1.4
Giorgio Incantalupo wrote: Hi Mark, it is show queues I use to see if phones are paused or not. The phones I'm using for tests are all SIP phones. Yes, what you are supposing could be right...Asterisk could see the phones as stuck. I'm still investigating, making test on my 1.4 box and I have noticed some other strange things about the phones. Some phones when normally used (I made a test making an outbound call) are seen as paused (In use) while other are marked as In Use only: (from Asterisk CLI): SIP/8 with penalty 1 (In use) has taken 1 calls (last was 3247 secs ago)(my phone) SIP/36 with penalty 1 (paused) (In use) has taken no calls yet(my test phone) The phones are the same model and have same sip.conf definition. The queues.conf definitions are the same for the two queues the phones are in. I do not know why queues show shows paused or not for similar phones. Can this be useful!?!? Giorgio The only way that a phone should become automatically paused is if the autopause option is set in queues.conf for the queue. There are ways through the dialplan and manager to manually pause a queue member, but there are no other ways for a member to become automatically paused. That being said, it could be that you have discovered some sort of bug in 1.4. When does this appear to happen? Does it happen randomly or is the situation reproduceable? Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with dial plan
Hi list,
Have installed trixbox and I am working with a fxo gateway to get fxo
calls to trixbox. I am using sip to send the calls from the gateway to
trixbox. I have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial plan below.
exten = 3000,1,dial(sip/3000)
exten= 3000,2,answer()
exten = 3000,3,congestion()
exten= 3000,4,hangup()
this works fine. But I when I put it in the form
exten = _3XXX,1,dial(sip/${EXTEN})
exten= _3XXX,2,answer()
exten =_3XXX,3,congestion()
exten= _3XXX,4,hangup()
the call goes into congestion and I get a busy tone. What could I be
doing wrong?
James
begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya
email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED]
title:Lead Consultant
tel;work:+254-722-490994
tel;home:+254-722-490994
tel;cell:+254-722-490994
url:www.agile.co.ke
version:2.1
end:vcard
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Re: [asterisk-users] [Posible Spam] asterisk-users Digest, Vol 48, Issue 59
Hola, Estoy de vacaciones hasta el 1 de Agosto. Para dar soporte sobre la centralita de telefonia: [EMAIL PROTECTED] Perdonen las molestias. Ruth Llaneza Lapausa - Tecnico de VoIP. [EMAIL PROTECTED] Tlf: 902 199 384 Mildmac SA – www.mildmac.es – [EMAIL PROTECTED] C/ Hnos. García Noblejas 41, 6ª planta. 28037 - Madrid Tlf: +34 91 501 33 02 Fax: +34 91 501 57 45 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Overlap dialing via SIP
Hi
I have set up an asterisk system which allows the use of Overlap Dialing from
SIP handsets. In order to do this I had to list the various patterns of numbers
which can be dialed in the UK. We also dial with a prefix of '9' for and outside
line so much of my dialplan looks like this :-
[084x]
exten = _9084,1,Macro(dialout-pstn)
[outbound-national]
exten = _90[1-2]X,1,Macro(dialout-pstn)
[087x]
exten = _9087,1,Macro(dialout-pstn)
[0906]
exten = _90906XXX,1,Macro(dialout-pstn)
...
I was able to download the mappings for 0800 numbers and other special ranges
from the ofcom website and I have incorporated these. For international dialing
I have not been able to find an easy way of doing this so I created the folling
contexts whcih make use of the WaitExten feature :-
[outbound-international]
exten = _900XX,1,Set(oldexten=${EXTEN})
exten = _900XX,2,Goto(international-number-length-check,s,1)
[international-number-length-check]
exten = s,1,Answer
exten = s,2,WaitExten(8)
exten = _X,1,Set(enddigits=${EXTEN})
exten = _X,2,NoOp(${TIMESTAMP} ok 13 digits - we dial ${oldexten}${enddigits})
exten = _X,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _X,4,Congestion()
exten = _X,104,Busy()
exten = _XX,1,Set(enddigits=${EXTEN})
exten = _XX,2,NoOp(${TIMESTAMP} ok 14 digits - we dial ${oldexten}${enddigits})
exten = _XX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _XX,4,Congestion()
exten = _XX,104,Busy()
exten = _XXX,1,Set(enddigits=${EXTEN})
exten = _XXX,2,NoOp(${TIMESTAMP} ok 15 digits - we dial
${oldexten}${enddigits})
exten = _XXX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _XXX,4,Congestion()
exten = _XXX,104,Busy()
exten = t,1,Dial(${OUTBOUNDTRUNK}/${oldexten})
exten = t,2,Congestion()
exten = t,102,Busy()
This works fairly well but I have noticed that occasionally the WaitExten
feature does
not seem to catch the first digits if they are dialed too quickly. It is almost
as if
there is a some sort of delay and the thirteenth digit is sometimes missed.
Can anyone suggest why WaitExten might be ocasionally missing a digit or can
anyone think
of a better way of doing this?
Thanks
Ben Thompson
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[asterisk-users] Asterisk Recording tools
Hello all I am looking for a recording tool for large environment, searching on the web I found that oreka is a great tool for this issue, anyone knows other tool or web gui to access to asterisk recordings? Anyone have installed successfully oreka recording tool? Thanks for any data. Cheers! Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Posible Spam] asterisk-users Digest, Vol 48, Issue 59
[EMAIL PROTECTED] schrieb: Estoy de vacaciones hasta el 1 de Agosto. Auto-responders should not reply to messages with any of the headers in following: Precedence: list or Precedence: bulk or List-* Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Recording tools
Gustavo, You may want to try out Druid (http://www.voiceroute.org) Open Source Edition which has free recording abilities for conference, queues, individual extensions controllable by the admin individual user. Druid Open Source Edition is free and open source. Ming On Tue, Jul 22, 2008 at 12:19 AM, Gustavo A Gonzalez [EMAIL PROTECTED] wrote: Hello all I am looking for a recording tool for large environment, searching on the web I found that oreka is a great tool for this issue, anyone knows other tool or web gui to access to asterisk recordings? Anyone have installed successfully oreka recording tool? Thanks for any data. Cheers! Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-877-242-3704 Office: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] -- Meet us at OSCON 2008, 21-25 Jul 2008, Oregon Convention Center, Booth 221 http://druidoscon.eventbrite.com Meet us at LinuxWorld 2008, 4-7 Aug 2008, Moscone Center, San Francisco, Booth 1626 http://druidlinuxworld.eventbrite.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] increase ring time out
I need to increase the ringing timeout on the AA50 appliance. How do I accomplish this? I need the phones to ring a bit more before the caller gets to the voicemail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Recording tools
Ask George Bush what he uses! On Mon, Jul 21, 2008 at 11:19 PM, Gustavo A Gonzalez [EMAIL PROTECTED] wrote: Hello all I am looking for a recording tool for large environment, searching on the web I found that oreka is a great tool for this issue, anyone knows other tool or web gui to access to asterisk recordings? Anyone have installed successfully oreka recording tool? Thanks for any data. Cheers! *Gustavo A. González* Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working. I am getting a SIP/401 Unauthorized error and then a SIP/404 error. I changed nothing in the configs. Is there a particular parameter needed for 1.6 that 1.4 did not care about? If I drop back to 1.4 it starts working again. Thanks Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo Issue
This is almost standard with voip calls. The echo-cancellation has to train up to the call parameters. Some hardware is better with it than others and you can try tweaking the value for the echo canceler up and down. What type hardware are you using - both phone and server? Hi, I have Astra 480i's and Snom M3's. I am using a SIP provider so I do not have any peripheral cards. I am on voip-wiki now reading about the echo canceller tuning, thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cascading Asterisk PBX
Hi to All, I have a PBX (MAINPBX) from a Telecomm Provider, which have the feature to transfer calls (Incoming call - Answer - FLASH - Dial Number to transfer - Answer - FLASH+4) and the call is transferred, but I have the need to implement an internal ACD using Asterisk as the PBX, the trunks connected to my Asterisk FXO ports are Extensions of my MAINPBX (ex., 5437, 5440 etc), all features work fine, but I have the need to make asterisk act as a normal telephone when transferring calls, I need to release the line (FXO port in my Asterisk) and make the transfer via the MAINPBX feature. Otherwise I will use 2 lines of my Asterisk PBX to make the transfer and it reduce the incoming lines available for my ACD. It's possible send the commands FLASH, FLASH+4 using the incoming line to my MAINPBX via Asterisk like a normal telephone? Thanks in Advance. Ricardo Melendez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Recording tools
Gustavo A Gonzalez wrote: Hello all I am looking for a recording tool for large environment, searching on the web I found that oreka is a great tool for this issue, anyone knows other tool or web gui to access to asterisk recordings? Anyone have installed successfully oreka recording tool? Thanks for any data. OrecX comes with a GUI. Now, I won't refrain from allegations of braindeath related to its design; it is some gargantuan JSP/servlet-driven monstrosity that could have been reproduced in probably 50 lines of PHP or Perl. I've never seen anything else that looks quite so much like a Java web development fanboy's work on a rooftop, in a snowstorm, to which - along with good software development practice - he was oblivious because he was loaded up on meth. But it does work, you might say. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems w/Asterisk Realtime + MySQL + SIP
[Jul 21 15:28:21] DEBUG[2028] chan_sip.c: Received REGISTER (2) -
Command in SIP REGISTER
[Jul 21 15:28:21] DEBUG[2028] chan_sip.c: SIP message could not be
handled, bad request: ZjFhZjZlNmZmZjM3OWFlYzE0MGYzZDYwYzJmODAwNDg.
It looks like Asterisk is unhappy with the SIP REGISTER request coming
from your softphone for some reason. It's very strange that it's
occurring for two different softphones though.
Trun on SIP debugging by typing sip debug on your Asterisk console
and then post up the 4 SIP messages invloved in the register
transaction so we can take a look and spot why it could be getting
rejected.
Sure.
Here's what happens when kphone starts up:
==
--- SIP read from 192.168.0.25:5060 ---
REGISTER sip:192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C
CSeq: 35 REGISTER
To: Walter sip:[EMAIL PROTECTED]
Expires: 900
From: Walter sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
Content-Length: 0
User-Agent: kphone/4.2
Event: registration
Allow-Events: presence
Contact: Walter
sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE,
INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER
black*CLI
-
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.25 : 5060 (no NAT)
--- Transmitting (no NAT) to 192.168.0.25:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C;received=192.168.0.25
From: Walter sip:[EMAIL PROTECTED]
To: Walter sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 35 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
--- Transmitting (no NAT) to 192.168.0.25:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C;received=192.168.0.25
From: Walter sip:[EMAIL PROTECTED]
To: Walter sip:[EMAIL PROTECTED];tag=as59de1023
Call-ID: [EMAIL PROTECTED]
CSeq: 35 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7864265a
Content-Length: 0
Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in
32000 ms (Method: REGISTER)
==
Kphone prompts for a password, then the following occurs.
==
--- SIP read from 192.168.0.25:5060 ---
REGISTER sip:192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C
CSeq: 36 REGISTER
To: Walter sip:[EMAIL PROTECTED]
Authorization: Digest username=walter, realm=asterisk,
nonce=7864265a, uri=sip:192.168.0.2, cnonce=abcdefghi,
nc=0001, response=10a7024959390c04b4d09c708fac6130, opaque=,
algorithm=MD5
Expires: 900
From: Walter sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
Content-Length: 0
User-Agent: kphone/4.2
Event: registration
Allow-Events: presence
Contact: Walter
sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE,
INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER
-
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.25 : 5060 (no NAT)
--- Transmitting (no NAT) to 192.168.0.25:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C;received=192.168.0.25
From: Walter sip:[EMAIL PROTECTED]
To: Walter sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 36 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
--- Transmitting (no NAT) to 192.168.0.25:5060 ---
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C;received=192.168.0.25
From: Walter sip:[EMAIL PROTECTED]
To: Walter sip:[EMAIL PROTECTED];tag=as59de1023
Call-ID: [EMAIL PROTECTED]
CSeq: 36 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
[Jul 22 00:59:38] NOTICE[2414]: chan_sip.c:15049
handle_request_register: Registration from 'Walter
sip:[EMAIL PROTECTED]' failed for '192.168.0.25' - Wrong password
Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in
32000 ms (Method: REGISTER)
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER
==
Just to confirm, the password supplied was 'aaa'.
In MySQL md5secret = md5('aaa') and secret = 'aaa'.
Here's what happens with zoiper (one registration click only)...
==
--- SIP read from 192.168.0.25:5060 ---
REGISTER sip:192.168.0.2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
snip:5060;branch=z9hG4bK-d8754z-9eb0f4d56eb2c53a-1---d8754z-;rport
Max-Forwards: 70
Contact:
Re: [asterisk-users] Option 't' on DIal
Hi,
If 't' is set on Dial command, but then i set canreinvite=yes on the account
[100]
type=friend
host=dynamic
nat=yes
secret=100
canreinvite=yes --- if i set this
would asterisk still stay in the path?
regards,
nhadie
Mark Michelson wrote:
Nhadie wrote:
Hi,
I encountered something i can't understand. I've setup 2 extensions.
[100]
type=friend
host=dynamic
nat=yes
secret=100
[101]
type=friend
host=dynamic
nat=yes
secret=101
and on extensions.conf
exten = _1XX,1,Dial(SIP/${EXTEN}|30|t)
exten = _1XX,n,Hangup
This dial plan is ok, audio connects both ways.
but when i had a typo error, i forgot the 't' option, only one way audio
when i call, 't' option is used to transfer call how come it affected
the audio?
thank you in advanced
regards
nhadie
The 't' option is one that requires Asterisk to be in the media path of the
call
(so that Asterisk can tell when the transfer DTMF has been pressed). In order
to
stay in the path, SIP reinvites are disabled for the call. Without the 't'
option, Asterisk will send reinvites to the phones so that their media does
not
go through Asterisk at all.
In order to figure out why there is one-way audio, you would need to provide
a
sip debug of the call. Based on the fact that you have nat=yes for both SIP
friends, I'm guessing that there's some sort of NAT issue here, but I can't
be
certain.
Mark Michelson
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Re: [asterisk-users] Option 't' on DIal
Nhadie wrote: Hi, If 't' is set on Dial command, but then i set canreinvite=yes on the account [100] type=friend host=dynamic nat=yes secret=100 canreinvite=yes --- if i set this would asterisk still stay in the path? regards, nhadie Yes, the Dial option will override the sip.conf option. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Bridge Command/Event in 1.6?
Thanks Olle. How do I use it? What's the parameters??? Doug. - Original Message From: Johansson Olle E [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 20, 2008 1:36:24 AM Subject: Re: [asterisk-users] New Bridge Command/Event in 1.6? 20 jul 2008 kl. 02.55 skrev Douglas Garstang: I just downloaded Asterisk 1.6 beta 9 because I had read that there was a new bridge command. After looking through the doc/* documentation, I see no mention of a bridge application or AMI command. Does it exist? I am trying to take a bridged call, and redirect each to another destination, which I can do with the redirect() AMI command. After doing some dial plan processing, I would like to bridge them back together. How can I do this? The redirect command takes a channel and an extension as an argument, not another channel. Read the CHANGES file: * Added a Bridge action which allows you to bridge any two channels that are currently active on the system. The developer forgot to add documentation to doc/manager_1_1.txt. Adding doc would be helpful. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Recording tools
On Mon, Jul 21, 2008 at 01:36:10PM -0400, Alex Balashov wrote: OrecX comes with a GUI. Now, I won't refrain from allegations of braindeath related to its design; it is some gargantuan JSP/servlet-driven monstrosity that could have been reproduced in probably 50 lines of PHP or Perl. I've never seen anything else that looks quite so much like a Java web development fanboy's work on a rooftop, in a snowstorm, to which - along with good software development practice - he was oblivious because he was loaded up on meth. But it does work, you might say. Don't hold back, Alex. Tell us how you /really/ feel. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
Jerry Geis wrote: I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working. I am getting a SIP/401 Unauthorized error and then a SIP/404 error. I changed nothing in the configs. How are you getting SIP-related errors from Console/DSP? Posting a console log would be most helpful, as many people on the mailing list are not telepathic :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Recording tools
Jay R. Ashworth wrote: On Mon, Jul 21, 2008 at 01:36:10PM -0400, Alex Balashov wrote: OrecX comes with a GUI. Now, I won't refrain from allegations of braindeath related to its design; it is some gargantuan JSP/servlet-driven monstrosity that could have been reproduced in probably 50 lines of PHP or Perl. I've never seen anything else that looks quite so much like a Java web development fanboy's work on a rooftop, in a snowstorm, to which - along with good software development practice - he was oblivious because he was loaded up on meth. But it does work, you might say. Don't hold back, Alex. Don't worry, I won't. :-) -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
ow are you getting SIP-related errors from Console/DSP? Posting a console log would be most helpful, as many people on the mailing list are not telepathic :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) Kevin, below is the log your talking about. please note no configuration files were changed from 1.4 to 1.6, going back to 1.4 works again. Jerry -- Asterisk 1.6.0-beta9, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': == Found [0;37;40m[1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': [1;30;40m == [0;37;40mFound [0mConnected to Asterisk 1.6.0-beta9 currently running on ebox4300 (pid = 4877) ebox4300*CLI Verbosity is at least 5 [Kebox4300*CLI --- SIP read from UDP://192.168.1.8:5060 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK029ea409;rport From: Jerry Geis 204 sip:[EMAIL PROTECTED];tag=as7d1f7b71 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 Jul 2008 16:53:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 20475 20475 IN IP4 192.168.1.8 s=session c=IN IP4 192.168.1.8 t=0 0 m=audio 14322 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - �--- (14 headers 14 lines) --- � == Using SIP RTP CoS mark 5 � == Using SIP VRTP CoS mark 6 �Sending to 192.168.1.8 : 5060 (NAT) �Using INVITE request as basis request - [EMAIL PROTECTED] �No user '3175661677' in SIP users list �Found peer 'devcentos5x64_to_ebox4300' for '3175661677' from 192.168.1.8:5060 � --- Reliably Transmitting (no NAT) to 192.168.1.8:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK029ea409;received=192.168.1.8;rport=5060 From: Jerry Geis 204 sip:[EMAIL PROTECTED];tag=as7d1f7b71 To: sip:[EMAIL PROTECTED];tag=as324df4b6 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0-beta9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0e961d2a Content-Length: 0 �Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) � [Kebox4300*CLI --- SIP read from UDP://192.168.1.8:5060 --- ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK029ea409;rport From: Jerry Geis 204 sip:[EMAIL PROTECTED];tag=as7d1f7b71 To: sip:[EMAIL PROTECTED];tag=as324df4b6 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 - �--- (10 headers 0 lines) --- � --- SIP read from UDP://192.168.1.8:5060 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK6a460d62;rport From: Jerry Geis 204 sip:[EMAIL PROTECTED];tag=as7d1f7b71 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=devcentos5x64_to_ebox4300, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=0e961d2a, response=1a8e257ae008af4156b1f65be8d4d267 Date: Mon, 21 Jul 2008 16:53:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 20475 20476 IN IP4 192.168.1.8 s=session c=IN IP4 192.168.1.8 t=0 0 m=audio 14322 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - �--- (15 headers 14 lines) --- �Sending to 192.168.1.8 : 5060 (NAT) �Using INVITE request as basis request - [EMAIL PROTECTED] �No user '3175661677' in SIP users list �Found peer 'devcentos5x64_to_ebox4300' for '3175661677' from 192.168.1.8:5060 �Found RTP audio format 0 �Found RTP audio format 8 �Found RTP audio format 3 �Found RTP audio format 101 �Peer audio RTP is at port 192.168.1.8:14322 �Found audio description format PCMU for ID 0 �Found audio description format PCMA for ID 8 �Found audio description format GSM for ID 3 �Found audio
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
Jerry Geis wrote: �Looking for mediaport_audio_visual in smvoice-mediaport (domain 192.168.1.25) Do you have an extension called 'mediaport_audio_visual' in a context called 'smvoice-mediaport'? If so, can you post that context so we can see how it looks? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
Do you have an extension called 'mediaport_audio_visual' in a context called 'smvoice-mediaport'? If so, can you post that context so we can see how it looks? Kevin, I mentioned that 1.4 works - 1.6 did not, going back to 1.4 works again. Here are the pieces: my sip.conf has context pointing to smvoice-mediaport part of extensions.conf: [smvoice-mediaport] exten = public_address,1,Goto(smvoice-mediaport-public-address,s,1) #include /etc/asterisk/express.dnis.conf file /etc/asterisk/express.dnis.conf ; MMAUDIO : EBOX 4300 - exten = mediaport_audio_visual,1,Goto(smvoice-mediaport-audio-visual,s,1) ; MMAUDIO : EBOX 4300 - exten = 1054,1,Goto(smvoice-mediaport-audio-visual,s,1) Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo Issue
Hi Joseph - I have Astra 480i's and Snom M3's. I am using a SIP provider so I do not have any peripheral cards. I am on voip-wiki now reading about the echo canceller tuning, thanks! For your particular case, you're probably not going to find much useful info on the wiki about echo cancellation. The info there is about reducing echo when there is an analog-to-digital conversion (in other words, if you're connecting to PSTN lines somewhere). If you have echo on calls that go through your SIP provider, it is possible that they are not doing a very good job with echo cancellation. If the echo is exclusively on these calls, you'll probably want to call them to discuss this. If you have echo on calls between your Astra and/or Snom handsets, you may want check the gain settings on these devices. Reducing the gain would probably lessen the effect of the echo. You may also want to check if either of these phones is doing any AEC (acoustic echo cancellation), and if there are any AEC parameters that are adjustable. I don't have experience with either of these phones, so I can't give you direct info on how to do this, but I'm sure that at least Snom support can help you. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP Packets Going To Wrong IP Address
I have a user behind a firewall who's had no issues in the past connecting though his firewall. He's registered just fine. But when he places a call, a large number of them have no audio on either side of the connection. No one can hear him, he can't hear anyone as well. After a lot of poking around (and changing many settings) I noticed that Asterisk is communicating the RTP packets to an internal IP address. My server has no internal IP address, only an external address, so it's not like we're trying to route this anywhere else. As can be seen below, I've already identified the host as being behind a firewall and therefor to not trust packets from it. Anyone have a suggestion? Name/username HostDyn Nat ACL Port Status Realtime jfabriquer/jfabriquer 75.36.34.98 D N 55266OK (145 ms) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Sent RTP P2P packet to 192.168.1.64:38826 (type 00, len 000160) Asterisk SVN-branch-1.4-r118365 -- Nicholas Blasgen [EMAIL PROTECTED] 408.497.9796 (c) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Heavy Load Asterisk Array
Hi everybody! I'm have to install some Asterisks in heavy load scenario with a load balance schema. The question is not very technical nor how to do it. I jut want to know if any of you have ever done an installation like this. Let me be more precise: 10 Asterisk servers, 2 OpenSer servers. I don't care much about OpenSER, but it would be great to have some succesful or unsuccesful ones justo to one if it can be done or not. I don't want to use my client as an expriment because it is a very big one. I'll appreciate your help. Thanks in advance. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Asterisk User #299 Share your knowledge, use free software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heavy Load Asterisk Array
I have used the OpenSer dispatcher module to load the calls (hash by caller id) to a group of asterisk boxes (In my case, 2 servers). The Asterisk boxes both use ARA and MySQL Master/Master replication. In a case like yours, I think you can use MySQL cluster, and you can still use Dispatcher to balance the load. On Mon, Jul 21, 2008 at 5:22 PM, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everybody! I'm have to install some Asterisks in heavy load scenario with a load balance schema. The question is not very technical nor how to do it. I jut want to know if any of you have ever done an installation like this. Let me be more precise: 10 Asterisk servers, 2 OpenSer servers. I don't care much about OpenSER, but it would be great to have some succesful or unsuccesful ones justo to one if it can be done or not. I don't want to use my client as an expriment because it is a very big one. I'll appreciate your help. Thanks in advance. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Asterisk User #299 Share your knowledge, use free software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heavy Load Asterisk Array
We also have the similar setup, 2 ser server with heartbeat doing the load balance and 4 asterisk servers handling the media. Of course the data is on MySQL Cluster. Jai Rangi www.bingotelecom.com On Mon, Jul 21, 2008 at 5:13 PM, Edgar Guadamuz [EMAIL PROTECTED] wrote: I have used the OpenSer dispatcher module to load the calls (hash by caller id) to a group of asterisk boxes (In my case, 2 servers). The Asterisk boxes both use ARA and MySQL Master/Master replication. In a case like yours, I think you can use MySQL cluster, and you can still use Dispatcher to balance the load. On Mon, Jul 21, 2008 at 5:22 PM, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everybody! I'm have to install some Asterisks in heavy load scenario with a load balance schema. The question is not very technical nor how to do it. I jut want to know if any of you have ever done an installation like this. Let me be more precise: 10 Asterisk servers, 2 OpenSer servers. I don't care much about OpenSER, but it would be great to have some succesful or unsuccesful ones justo to one if it can be done or not. I don't want to use my client as an expriment because it is a very big one. I'll appreciate your help. Thanks in advance. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Asterisk User #299 Share your knowledge, use free software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automon=*, Dial(, , Ww), rfc2833, canreinvite=no, but...
On Fri, 2008-07-18 at 13:02 -0400, Bill Michaelson wrote: After much checking and puzzling, I cannot get my Polycom 601 to toggle call recording with my Asterisk 1.4.21.1. I can see this in the feature*.conf file set: automon=*1 and I can see a 'Ww' in the logged/traced call to dial(). Is the DYNAMIC_FEATURES variable set correctly? It's been my experience that not setting DYNAMIC_FEATURES is the number one problem people encounter with one-touch recording. Finally, it might be worth noting that the packet traces show three RFC2833 end events for each DTMF code pressed. This might be perfectly normal Perfectly normal... nothing to worry about there. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems w/Asterisk Realtime + MySQL + SIP
Hi, Try to delete whole column 'md5secret' from DB peers table. Leave only 'secret'. And try then. Regards, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Walter Stanish Sent: Monday, July 21, 2008 8:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems w/Asterisk Realtime + MySQL + SIP [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: Received REGISTER (2) - Command in SIP REGISTER [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: SIP message could not be handled, bad request: ZjFhZjZlNmZmZjM3OWFlYzE0MGYzZDYwYzJmODAwNDg. It looks like Asterisk is unhappy with the SIP REGISTER request coming from your softphone for some reason. It's very strange that it's occurring for two different softphones though. Trun on SIP debugging by typing sip debug on your Asterisk console and then post up the 4 SIP messages invloved in the register transaction so we can take a look and spot why it could be getting rejected. Sure. Here's what happens when kphone starts up: == --- SIP read from 192.168.0.25:5060 --- REGISTER sip:192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C CSeq: 35 REGISTER To: Walter sip:[EMAIL PROTECTED] Expires: 900 From: Walter sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.2 Event: registration Allow-Events: presence Contact: Walter sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER black*CLI - --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.25 : 5060 (no NAT) --- Transmitting (no NAT) to 192.168.0.25:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C;received=192.168.0.25 From: Walter sip:[EMAIL PROTECTED] To: Walter sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 35 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.25:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C;received=192.168.0.25 From: Walter sip:[EMAIL PROTECTED] To: Walter sip:[EMAIL PROTECTED];tag=as59de1023 Call-ID: [EMAIL PROTECTED] CSeq: 35 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7864265a Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) == Kphone prompts for a password, then the following occurs. == --- SIP read from 192.168.0.25:5060 --- REGISTER sip:192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C CSeq: 36 REGISTER To: Walter sip:[EMAIL PROTECTED] Authorization: Digest username=walter, realm=asterisk, nonce=7864265a, uri=sip:192.168.0.2, cnonce=abcdefghi, nc=0001, response=10a7024959390c04b4d09c708fac6130, opaque=, algorithm=MD5 Expires: 900 From: Walter sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.2 Event: registration Allow-Events: presence Contact: Walter sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER - --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.25 : 5060 (no NAT) --- Transmitting (no NAT) to 192.168.0.25:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C;received=192.168.0.25 From: Walter sip:[EMAIL PROTECTED] To: Walter sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 36 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.25:5060 --- SIP/2.0 403 Forbidden (Bad auth) Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C;received=192.168.0.25 From: Walter sip:[EMAIL PROTECTED] To: Walter sip:[EMAIL PROTECTED];tag=as59de1023 Call-ID: [EMAIL PROTECTED] CSeq: 36 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 [Jul 22 00:59:38] NOTICE[2414]: chan_sip.c:15049 handle_request_register: Registration from 'Walter sip:[EMAIL PROTECTED]' failed for '192.168.0.25' - Wrong password Scheduling destruction of SIP dialog '[EMAIL
Re: [asterisk-users] RTP Packets Going To Wrong IP Address
On Mon, 21 Jul 2008, Nicholas Blasgen wrote: I have a user behind a firewall who's had no issues in the past connecting though his firewall. He's registered just fine. But when he places a call, a large number of them have no audio on either side of the connection. No one can hear him, he can't hear anyone as well. After a lot of poking around (and changing many settings) I noticed that Asterisk is communicating the RTP packets to an internal IP address. My server has no internal IP address, only an external address, so it's not like we're trying to route this anywhere else. As can be seen below, I've already identified the host as being behind a firewall and therefor to not trust packets from it. Anyone have a suggestion? Ask them if they're replaced their router recently? If so, see if it's got a broken SIP ALG... (Some Draytek, Cisco, Zyxel for example) Get them to remove all port-forwarding on their firewall, remove all fancy port/ip address settings on their phone and use a STUN server. If they are using STUN, make sure it's working. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help With dial plan
Hi list,
Have installed trixbox and I am working with a fxo gateway to get fxo
calls to trixbox. I am using sip to send the calls from the gateway to
trixbox. I have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial plan below.
exten = 3000,1,dial(sip/3000)
exten= 3000,2,answer()
exten = 3000,3,congestion()
exten= 3000,4,hangup()
this works fine. But I when I put it in the form
exten = _3XXX,1,dial(sip/${EXTEN})
exten= _3XXX,2,answer()
exten =_3XXX,3,congestion()
exten= _3XXX,4,hangup()
the call goes into congestion and I get a busy tone. What could I be
doing wrong?
James
begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya
email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED]
title:Lead Consultant
tel;work:+254-722-490994
tel;home:+254-722-490994
tel;cell:+254-722-490994
url:www.agile.co.ke
version:2.1
end:vcard
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