Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
I think it can't hurt to try a different release. Let me know how it goes. Thanks Igor. I just upgraded zaptel to 1.4.11. However, I am still seeing red in the alarm in zttool and the LED on port 1 also shows red. --- cat /proc/zaptel/1 is also showing Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 RED 1 TE4/0/1/1 Clear RED 2 TE4/0/1/2 Clear RED Zaptel started up fine and dmesg below does not show the error message. I am just wondering whether this China E1 could be using MFC R/2? How do I know it is? Stopped TE4XXP, Turned off DMA TE4XXP: Disabling interrupts since there are no active spans Unregistered Tormenta2 Registered Tormenta2 PCI Found TE4XXP at base address fc4ffc00, remapped to f89f4c00 TE4XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x35fe0400 Reg 1: 0x35fe Reg 2: 0x Reg 3: 0x Reg 4: 0x0101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1100 Reg 8: 0x010200ff Reg 9: 0x00fd0001 Reg 10: 0x004a TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE410P (4th Gen) usbcore: registered new driver wcusb Wildcard USB FXS Interface driver registered About to enter spanconfig! Done with spanconfig! About to enter startup! TE4XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 1: Primary Sync Source VPM400: Not Present VPM450: echo cancellation for 128 channels VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) Completed startup! ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
Wow - that's nasty. Almost like a broken card or MB. Ouch. Should you call the supplier of the card and ask them about warranty? PaulH Thanks Paul. The TE412P card is fine and the zaptel error in dmesg is fixed by 1.4.11. However, the red alarms are still there. ___ -- 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] OT - How to test tftp for phones provisioning
2008/7/28 Drew Gibson [EMAIL PROTECTED] Olivier, also check that you don't have any firewalls in the way, especially if there is nothing in the logs. Turn them off for testing, on both server and CLIENT. I found out about the client f/w blocking tftp the hard way! :-) It's good to be aware of that (though I didn't have this issue) Thanks ! regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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 ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
Wow - that's nasty. Almost like a broken card or MB. Ouch. Should you call the supplier of the card and ask them about warranty? PaulH Lee, John (Sydney) wrote: This time, I am trying to remotely install Asterisk in China. I was told that an E1 line has been installed and so I plug it into port 1 of a TE412P. On the box, first of all, I just installed Zaptel 1.4.10.1. # service zaptel restart Unloading zaptel hardware drivers:ERROR: Module zaptel is in use . Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. Running ztcfg: [ OK ] # vi zaptel.conf [...] span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 *** However, I received a red alarm in zttool and the LED on the TE412P card is also red. *** I have made sure that the jumper is closed for port 1 on the TE412P card and so it could not be the jumper problem. ### Because this is the first time I install Asterisk in China and I was wondering if their E1 is different from the Euro E1. ### However, I went into dmesg and I discovered the following. ### Could it really be a zaptel bug? I saw on a similar few on the digium bug list but I cannot be 100% sure. Any thoughts? About to enter spanconfig! Done with spanconfig! Registered tone zone 33 (China) About to enter startup! TE4XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 128 channels BUG: soft lockup - CPU#2 stuck for 16s! [ztcfg:4681] Pid: 4681, comm:ztcfg EIP: 0060:[f8cba1df] CPU: 2 EIP is at init_vpm450m+0x32d/0x34a [wct4xxp] EFLAGS: 0286Tainted: G (2.6.18-92.1.6.el5 #1) EAX: EBX: f76ae8f0 ECX: 0019 EDX: ESI: f7997400 EDI: 0286 EBP: f76838c0 DS: 007b ES: 007b CR0: 8005003b CR2: b7f01007 CR3: 37bc1000 CR4: 06d0 [f8ca1b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8ca5ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042609c] release_console_sem+0x17e/0x1b8 [c046d53a] cache_alloc_refill+0x14b/0x450 [f8956f61] zt_ioctl+0x273/0x144f [zaptel] [c04d7d45] generic_make_request+0x248/0x258 [c045ae3c] __do_page_cache_readahead+0x69/0x1c6 [c0484a5b] __d_lookup+0x98/0xdb [c047c110] do_lookup+0x53/0x166 [c047e7e4] do_path_lookup+0x20e/0x25e [c047c389] permission+0xa2/0xb5 [c04e2d06] kobject_get+0xf/0x13 [c046f7fa] __dentry_open+0xea/0x1ab [c046f91f] nameidata_to_filp+0x19/0x28 [c046f959] do_filp_open+0x2b/0x31 [c048029b] do_ioctl+0x47/0x5d [c04804fb] vfs_ioctl+0x24a/0x25c [c0471bbe] __fput+0x13f/0x167 [c0480555] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) Completed startup! ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
On Tue, Jul 29, 2008 at 01:48:26PM +1000, Lee, John (Sydney) wrote: On the box, first of all, I just installed Zaptel 1.4.10.1. [..] BUG: soft lockup - CPU#2 stuck for 16s! [ztcfg:4681] Hi, just for (all of) you to know this is a known bug of zaptel 1.4.11, the firmware upload procedure is taking some time, operating like a freeze during the process, so this message appears. But this isn't a real problem, as it doesn't have any consequences appart from the message. -- benoit ___ -- 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] Need help with implementing prepaid in asterisk
Hi all I am trying to implement a prepaid dialing system on our asterisk box. I however have a few questions I need to ask. I have written a simple script in php to do all the billing. 1. What do I need to user to cut off the users in mid call 2. What do I need to insert into my dialplan to deny a user to call if you need any config files I will send them, seeing as I dont know what files to send. If you can point me to a howto I will be more gratefull. Regards Ian ___ -- 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] Outgoing calls
Hello list, How could I limit the outgoing calls for one trunks easily? Thanks VoipCrazy ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
Hi, just for (all of) you to know this is a known bug of zaptel 1.4.11, the firmware upload procedure is taking some time, operating like a freeze during the process, so this message appears. But this isn't a real problem, as it doesn't have any consequences appart from the message. Thanks benoit You are right. This was not the cause of the red alarm. Is there anywhere in the box I can look around to find out the cause of the red alarm? ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
On Tue, Jul 29, 2008 at 12:04:43AM -0500, Tilghman Lesher wrote: On Monday 28 July 2008 22:48:26 Lee, John (Sydney) wrote: This time, I am trying to remotely install Asterisk in China. I was told that an E1 line has been installed and so I plug it into port 1 of a TE412P. Are you sure that they're plugged into port 1 and not port 4? It is a rather common mistake to believe that the port numbers start at the bottom of the card and not at the top. The test for that is simple: head -n 1 /proc/zaptel/* Let's look at all four spans. Not just the first one. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
The test for that is simple: head -n 1 /proc/zaptel/* Let's look at all four spans. Not just the first one. Thanks Tzafrir. # head -n 1 /proc/zaptel/* == /proc/zaptel/1 == Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED == /proc/zaptel/2 == Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 == /proc/zaptel/3 == Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 == /proc/zaptel/4 == Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 So I am quite sure that port 1 is plugged in properly. As I am dealing with telecom in China, I think I might have stepped onto the MFC R/2 bombshell but I have no idea whether the signalling is ISDN or R2. I tried the suggestion on http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 but the red alarm is still on. If it is really R2, then maybe I need to buy an E100P card instead of TE412P. Any thoughts? ___ -- 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] Need help with implementing prepaid in asterisk
You can calculate users remaining minutes according to his remaining balance and then set the Absolute timeout for his every outgoing call using the Timeout(absolute) = X http://www.voip-info.org/wiki/index.php?page=Asterisk+func+timeoutvariable. This will hangup the call after X seconds And if there users balance is in negative or zero then just bypass the dial statement in your dialplan and land on Hangup or Playback before hangup to let the user know why his call is not being connected. All of these changes you have to make in extensions.conf On Tue, Jul 29, 2008 at 11:59 AM, Ian Coetzee [EMAIL PROTECTED]wrote: Hi all I am trying to implement a prepaid dialing system on our asterisk box. I however have a few questions I need to ask. I have written a simple script in php to do all the billing. 1. What do I need to user to cut off the users in mid call 2. What do I need to insert into my dialplan to deny a user to call if you need any config files I will send them, seeing as I dont know what files to send. If you can point me to a howto I will be more gratefull. Regards Ian ___ -- 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 -- Best Regards Rizwan Hisham ___ -- 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] Outgoing calls
try put calls into groups using GROUP() function and check call limit with GROUP_COUNT() voip crazy wrote: Hello list, How could I limit the outgoing calls for one trunks easily? Thanks VoipCrazy ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
On Tue, Jul 29, 2008 at 05:55:05PM +1000, Lee, John (Sydney) wrote: The test for that is simple: head -n 1 /proc/zaptel/* Let's look at all four spans. Not just the first one. Thanks Tzafrir. # head -n 1 /proc/zaptel/* == /proc/zaptel/1 == Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED It is RED. It means that no either there is no physical connection to the remote side, or otherwise both sides don't agree on the definitions. == /proc/zaptel/2 == Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 == /proc/zaptel/3 == Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 == /proc/zaptel/4 == Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 Those three were not configuerd (have no span= entries in zaptel.conf, I guess) so we have no idea if anything is plugged into them. So I am quite sure that port 1 is plugged in properly. As I am dealing with telecom in China, I think I might have stepped onto the MFC R/2 bombshell but I have no idea whether the signalling is ISDN or R2. Or, to phrase it in layer 1 terms: CAS (for MFC/R2) or CCS (for ISDN)? We're staying in layer 1 for the moment because the issue is the RED alarm. I tried the suggestion on http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 but the red alarm is still on. An output of 'head -n 1 /proc/zaptel/*' after the fact might help us help you. Did you re-run ztcfg after editing zaptel.conf ? If it is really R2, then maybe I need to buy an E100P card instead of TE412P. The TE412P supports R2 just as well. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
Hi John, Good to see a fellow Sydney-sider setting up Asterisk in China! Which city are you in? Our setup is in Kunming, Yunnan province. I've managed to get a working setup through China Netcom with a two-port Sangoma card. I'd email you all the settings / versions I'm using right now but I'm currently out of the country (in Penang, Malaysia), if you would like me to do so send me an email off-list and I'll try to get them for you. Didn't read the whole thread but one thing I'd check if it hasn't been mentioned yet is that the lights at the back of the card (if present) are showing OK, and the same for the cards on the router you're connected to. If both are OK, try changing all of your signalling settings. It took us awhile to hit on the right ones. 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
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
I tried the suggestion on http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 but the red alarm is still on. An output of 'head -n 1 /proc/zaptel/*' after the fact might help us help you. Did you re-run ztcfg after editing zaptel.conf ? Thanks Tzafrir (zaptel.conf) span=1,1,0,cas,hdb3 cas=1-15: cas=17-31: dchan=16 # service zaptel restart Unloading zaptel hardware drivers:ERROR: Module zaptel is in use . Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. Running ztcfg: [ OK ] # head -n 1 /proc/zaptel/* == /proc/zaptel/1 == Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED == /proc/zaptel/2 == Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 == /proc/zaptel/3 == Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 == /proc/zaptel/4 == Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # ztcfg -v Zaptel Version: 1.4.11 Echo Canceller: MG2 Configuration == SPAN 1: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: CAS / User (Default) (Slaves: 01) Channel 02: CAS / User (Default) (Slaves: 02) [...] Channel 15: CAS / User (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: CAS / User (Default) (Slaves: 17) [...] Channel 31: CAS / User (Default) (Slaves: 31) 31 channels to configure. # cat /proc/zaptel/1 Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED 1 TE4/0/1/1 CAS RED 2 TE4/0/1/2 CAS RED [...] Any thoughts? ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
On Tue, Jul 29, 2008 at 10:31:43PM +1000, Lee, John (Sydney) wrote: I tried the suggestion on http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 but the red alarm is still on. An output of 'head -n 1 /proc/zaptel/*' after the fact might help us help you. Did you re-run ztcfg after editing zaptel.conf ? Thanks Tzafrir (zaptel.conf) span=1,1,0,cas,hdb3 cas=1-15: cas=17-31: dchan=16 # service zaptel restart Unloading zaptel hardware drivers:ERROR: Module zaptel is in use . Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. Running ztcfg: [ OK ] # head -n 1 /proc/zaptel/* == /proc/zaptel/1 == Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED == /proc/zaptel/2 == Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 == /proc/zaptel/3 == Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 == /proc/zaptel/4 == Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # ztcfg -v Zaptel Version: 1.4.11 Echo Canceller: MG2 Configuration == SPAN 1: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: CAS / User (Default) (Slaves: 01) Channel 02: CAS / User (Default) (Slaves: 02) [...] Channel 15: CAS / User (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: CAS / User (Default) (Slaves: 17) [...] Channel 31: CAS / User (Default) (Slaves: 31) 31 channels to configure. # cat /proc/zaptel/1 Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED 1 TE4/0/1/1 CAS RED 2 TE4/0/1/2 CAS RED [...] Any thoughts? Yes, to try port 4 again. Backup the existing zaptel.conf and run genzaptelconf (no need to unload / reload any modules). What is the output of 'head -n 1 /proc/zaptel/*' after that? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] asterisk stops sending qualify
hi, i have problem with asterisk 1.4.20.1 (kernel 2.6.25.10, centos5, ztdummy+hrtimers) after some random time, asterisk stops sending qualify (monitored by wireshark) to peer (phone) before i'll go to bugs.digium.com is there someone with similar problem? thanks --- Marek Cervenka === ___ -- 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] Azurn International
I haven't seen any demo's of the product yet but looks like an interesting tool coming out of an Australian company called Azurn International. Australian technology company Azurn International has developed a proprietary online advertising marketing tool which will be initially implemented by ASX listed call center company UCMS. The Azurn Merlin click-to-connect technology allows website visitors to contact a customer service person via voice or video by clicking an icon on a webpage or email. The web based connection is free of charge to the customer, who can choose to be connected via their Internet phone, mobile or landline. The service aims to convert more web visitors into sales by capturing potential customers during their initial interest. Azurn International CEO Ananda Rao, says Merlin's ability to provide customers with human interaction in real-time and at no cost to the customer, will deliver greater ROI. Another feature of Merlin is that once connected via the phone, businesses can also share information and collaborate with the caller by exchanging relevant documents, forms and contracts over the webpage while the caller is connected. Anyone know anything about them? Or can you confirm they are using Asterisk? One dissapointing point about their website is in the first two paragraphs they mention the words proprietary and patenteda turn off for anyone who has been around the block more than a few times. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (New York) +61-2-9016-5642 (Sydney) http://www.Cognation.net http://www.Cognation.net/profile ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
Hi John, In China you will generally get either MFC/R2 or EuroISDN. MFC/R2 is much more common. The only oddity with EuroISDN is that it often provided without CRC4. That doesn't make a lot of sense, but there it is. MFC/R2 seems to be universally provided without CRC4 in China. You said you are sure the card is OK. How did you determine that? Have you tried a loopback cable between two its its ports? Are you sure the cable you have used is OK? Regards, Steve Lee, John (Sydney) wrote: I tried the suggestion on http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 but the red alarm is still on. An output of 'head -n 1 /proc/zaptel/*' after the fact might help us help you. Did you re-run ztcfg after editing zaptel.conf ? Thanks Tzafrir (zaptel.conf) span=1,1,0,cas,hdb3 cas=1-15: cas=17-31: dchan=16 # service zaptel restart Unloading zaptel hardware drivers:ERROR: Module zaptel is in use . Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. Running ztcfg: [ OK ] # head -n 1 /proc/zaptel/* == /proc/zaptel/1 == Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED == /proc/zaptel/2 == Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 == /proc/zaptel/3 == Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 == /proc/zaptel/4 == Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # ztcfg -v Zaptel Version: 1.4.11 Echo Canceller: MG2 Configuration == SPAN 1: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: CAS / User (Default) (Slaves: 01) Channel 02: CAS / User (Default) (Slaves: 02) [...] Channel 15: CAS / User (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: CAS / User (Default) (Slaves: 17) [...] Channel 31: CAS / User (Default) (Slaves: 31) 31 channels to configure. # cat /proc/zaptel/1 Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED 1 TE4/0/1/1 CAS RED 2 TE4/0/1/2 CAS RED [...] Any thoughts? ___ -- 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] asterisk+ fax-to-mail
Hello,
I have asterisk 1.2.7 and I would like to install the fax-to-mail.
I already installed spandsp (2.25), app_rxfax, app_txfax and
app_makefile.patch , I rebuilt Asterisk but when I send a fax, I dont
receive it.
I have this:
-- Executing Set(SIP/0172292965-726e, LANGUAGE()=fr) in new stack
-- Executing SetVar(SIP/0172292965-726e, FaxID=0172292965) in new
stack
-- Executing Goto(SIP/0172292965-726e, fax|0172292965|1) in new
stack
-- Goto (fax,0172292965,1)
-- Executing Macro(SIP/0172292965-726e, faxreceive) in new stack
-- Executing SetVar(SIP/0172292965-726e,
FAXFILE=/var/spool/asterisk/fax/1217324795.1.tif) in new stack
-- Executing DBget(SIP/0172292965-726e,
EMAILADDR=extensionemail/0172292965) in new stack
-- DBget: varname=EMAILADDR, family=extensionemail, key=0172292965
-- DBget: Value not found in database.
-- Executing SetVar(SIP/0172292965-726e,
[EMAIL PROTECTED]) in new stack
-- Executing Goto(SIP/0172292965-726e, 3) in new stack
-- Goto (macro-faxreceive,s,3)
-- Executing RxFAX(SIP/0172292965-726e,
/var/spool/asterisk/fax/1217324795.1.tif) in new stack
-- Executing System(SIP/0172292965-726e, /usr/sbin/mailfax
/var/spool/asterisk/fax/1217324795.1.tif [EMAIL PROTECTED] 0146446439) in
new stack
== Spawn extension (fax, h, 1) exited non-zero on 'SIP/0172292965-726e'
My file extensions.conf is configured like this:
[macro-faxreceive]
exten = s,1,Answer()
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(3)
I hope somebody could help me !
Regards ,
Nadjia.
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[asterisk-users] Asterisk SIP configuration
Hello, I'm trying to setup an instance of Asterisk to use as a SIP conference server only. And I'm using OpenSER as SIP Registrar/Proxy. On asterisk's sip.conf I created a user [EMAIL PROTECTED] (sipdomain == server name), that registers to OpenSER when asterisk is started. The user conference refers to context conference-context. And, on the extensions.conf I've defined the context called conference-context, with an extension named conference. From my SIP client, I call sip:[EMAIL PROTECTED] And get a 404 Not found response from asterisk. On asterisk's log I see messages like: Looking for conference on conference-context (domain serverIP) And: Call from 'conference' to extension 'conference' rejected because extension not found. Does anyone have an ideia of why I'm getting that message? Why does asterisk seem to be using domain == serverIP, instead of domain == servername? Is that correct the behavior? Or I may have something missing on my configuration? I suspect the problem is more likely be in the sip.conf file, but I can't see what's wrong/missing. I'm using app_conference. But I don't think this matters for now, because my first line of the conference extension calls Log(). And, as I don't see my log message printed, I assume asterisk didn't even start processing the commands defined for the conference extension. Thanks, Anderson Luiz Brunozi ___ -- 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] interactive IVR
Greetings.. My client is starting a new business and they required a strange thing.. they want an IVR system that can be integrated with some competetion.. the scenario is he folowing the caller VoIP Provided will reach the system.. the IVR picks up and welcomes them .. multiple language supports with multiple choices.. the caller piks his language and then they enter the competetion.. they hear a random question with multiple choices if he picks the right choice which is randomly set moves to the next level and congratultes him for the winning.. if he choses the wrong choice .. it tells him that he picked the wrong answer and asks if he wants to go farther and pick another question.. and when he finishes and wins .. they ask him to recors his details namephone number to contact him later for the price.. is asterisk capable of fulfilling that scenario? if not.. are there any other options? companies? best regards Tarek Sawah Integrated Digital Systems _ Use video conversation to talk face-to-face with Windows Live Messenger. http://www.windowslive.com/messenger/connect_your_way.html?ocid=TXT_TAGLM_WL_Refresh_messenger_video_072008___ -- 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] Newbie in China: Red alaram in Zaptel for E1
If you're interested, our working circuit in Chengdu with a China Telecom PRI is configured as: /etc/zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = cn defaultzone=cn /etc/asterisk/zapata.conf (just including the pertinent lines): switchtype=euroisdn signalling=pri_cpe group = 1 channel=1-15,17-31 Chris ___ Chris Hoff Telecommunications Administrator SEI LLC Voice +1 701 298 8865 Ext 2189 Mobile +1 701 361 5976 Fax +1 701 298 8860 Email [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Tuesday, July 29, 2008 8:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1 Hi John, In China you will generally get either MFC/R2 or EuroISDN. MFC/R2 is much more common. The only oddity with EuroISDN is that it often provided without CRC4. That doesn't make a lot of sense, but there it is. MFC/R2 seems to be universally provided without CRC4 in China. You said you are sure the card is OK. How did you determine that? Have you tried a loopback cable between two its its ports? Are you sure the cable you have used is OK? Regards, Steve Lee, John (Sydney) wrote: I tried the suggestion on http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 but the red alarm is still on. An output of 'head -n 1 /proc/zaptel/*' after the fact might help us help you. Did you re-run ztcfg after editing zaptel.conf ? Thanks Tzafrir (zaptel.conf) span=1,1,0,cas,hdb3 cas=1-15: cas=17-31: dchan=16 # service zaptel restart Unloading zaptel hardware drivers:ERROR: Module zaptel is in use . Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. Running ztcfg: [ OK ] # head -n 1 /proc/zaptel/* == /proc/zaptel/1 == Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED == /proc/zaptel/2 == Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 == /proc/zaptel/3 == Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 == /proc/zaptel/4 == Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # ztcfg -v Zaptel Version: 1.4.11 Echo Canceller: MG2 Configuration == SPAN 1: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: CAS / User (Default) (Slaves: 01) Channel 02: CAS / User (Default) (Slaves: 02) [...] Channel 15: CAS / User (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: CAS / User (Default) (Slaves: 17) [...] Channel 31: CAS / User (Default) (Slaves: 31) 31 channels to configure. # cat /proc/zaptel/1 Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED 1 TE4/0/1/1 CAS RED 2 TE4/0/1/2 CAS RED [...] Any thoughts? ___ -- 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] Cisco vs Asterisk
Am Freitag, 25. Juli 2008 18:54 schrieb Al Baker: This is simply NOT TRUE and shows a complete lack of understanding of modern software development. CISCO software is developed in a CMM environment. It has a formal test methodology and uses Automated Testing on EACH new release to ensure that 100% of the software that functioned in the Last Release, actually works in this release. This is simply NOT TRUE and shows a complete lack of understanding of software testing. Even if a company (or OS developer) implements automatic tests that cover each and every existing functionality this does NOT automatically ensure, that the existing functionalty works together well with new features. Nor does it ensure that existing code does work as intended under NEW circumstances. Testing (and I mean ANY form of testing, be it automatic or manual) can NEVER ensure the abscence of bugs! Additionally: Companies will never even atempt to find any bug. They will always only try to find as many bugs necessary to ensure that the cost of maintenance, bug-fixing, compensations and possibly loss of prestige does not exceed the cost of testing. Anything else would be financial loss. -- Stephan Weinberger ___ -- 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] interactive IVR
Yes. On Tue, Jul 29, 2008 at 10:17 AM, Tariq .. [EMAIL PROTECTED] wrote: Greetings.. My client is starting a new business and they required a strange thing.. they want an IVR system that can be integrated with some competetion.. the scenario is he folowing the caller VoIP Provided will reach the system.. the IVR picks up and welcomes them .. multiple language supports with multiple choices.. the caller piks his language and then they enter the competetion.. they hear a random question with multiple choices if he picks the right choice which is randomly set moves to the next level and congratultes him for the winning.. if he choses the wrong choice .. it tells him that he picked the wrong answer and asks if he wants to go farther and pick another question.. and when he finishes and wins .. they ask him to recors his details namephone number to contact him later for the price.. is asterisk capable of fulfilling that scenario? if not.. are there any other options? companies? best regards Tarek Sawah Integrated Digital Systems Use video conversation to talk face-to-face with Windows Live Messenger. Get started. ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
On Tue, Jul 29, 2008 at 01:56:23PM +0300, Tzafrir Cohen wrote: An output of 'head -n 1 /proc/zaptel/*' after the fact might help us help you. Did you re-run ztcfg after editing zaptel.conf ? On a sidebar, let me suggest head -1q; it's neater. 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] interactive IVR
On 29 Jul 2008, at 15:31, Steve Totaro wrote: Yes. Beat me to it ;) ___ -- 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 SIP configuration
Anderson Luiz Brunozi wrote: Hello, I'm trying to setup an instance of Asterisk to use as a SIP conference server only. And I'm using OpenSER as SIP Registrar/Proxy. On asterisk's sip.conf I created a user [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (sipdomain == server name), that registers to OpenSER when asterisk is started. The user conference refers to context conference-context. And, on the extensions.conf I've defined the context called conference-context, with an extension named conference. From my SIP client, I call sip:[EMAIL PROTECTED] And get a 404 Not found response from asterisk. On asterisk's log I see messages like: Looking for conference on conference-context (domain serverIP) And: Call from 'conference' to extension 'conference' rejected because extension not found. Does anyone have an ideia of why I'm getting that message? Why does asterisk seem to be using domain == serverIP, instead of domain == servername? Is that correct the behavior? Or I may have something missing on my configuration? I suspect the problem is more likely be in the sip.conf file, but I can't see what's wrong/missing. I'm using app_conference. But I don't think this matters for now, because my first line of the conference extension calls Log(). And, as I don't see my log message printed, I assume asterisk didn't even start processing the commands defined for the conference extension. Thanks, Anderson Luiz Brunozi ___ -- 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 Please send the relevant portion of your extensions.conf, as that is where the problem is -- Sherwood McGowan VoIP / Telecom Solutions [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] interactive IVR
Tariq .. wrote: Greetings.. My client is starting a new business and they required a strange thing.. they want an IVR system that can be integrated with some competetion.. the scenario is he folowing the caller VoIP Provided will reach the system.. the IVR picks up and welcomes them .. multiple language supports with multiple choices.. the caller piks his language and then they enter the competetion.. they hear a random question with multiple choices if he picks the right choice which is randomly set moves to the next level and congratultes him for the winning.. if he choses the wrong choice .. it tells him that he picked the wrong answer and asks if he wants to go farther and pick another question.. and when he finishes and wins .. they ask him to recors his details namephone number to contact him later for the price.. is asterisk capable of fulfilling that scenario? if not.. are there any other options? companies? best regards Tarek Sawah Integrated Digital Systems Use video conversation to talk face-to-face with Windows Live Messenger. Get started. http://www.windowslive.com/messenger/connect_your_way.html?ocid=TXT_TAGLM_WL_Refresh_messenger_video_072008 ___ -- 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 Yes, I know I could accomplish this using the dialplan and MySQL -- Sherwood McGowan VoIP / Telecom Solutions [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] Asterisk SIP configuration
Hi, Sherwood, Thanks for your reply. Here are my sip.conf and extensions.conf files. - begin extension.conf - [general] static=yes writeprotect=no autofallthrough=yes [globals] [conference-context] exten = s,1,Log(VERBOSE|Enter conference-context - extension s.) exten = s,2,Goto(conference,1) exten = conference,1,Log(VERBOSE|Enter conference-context.) exten = conference,2,Conference(1234/S/1) exten = conference,3,Hangup() - end extension.conf - - begin sip.conf - [general] bindport=6060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw autocreatepeer=yes autodomain=yes context=conference-context jbenable=yes jbmaxsize=200 jbimpl=adaptive jblog=yes fromdomain=dmd77 realm=dmd77 register = [EMAIL PROTECTED]:[EMAIL PROTECTED]/conference rtpkeepalive=5 insecure=invite sipdebug=yes nat=yes qualify=yes canreinvite=no [conference] type=friend nat=yes username=conference secret=conference canreinvite=no host=dmd77 context=conference-context fromdomain=dmd77 [guest] type=friend nat=yes host=dynamic canreinvite=no context=conference-context - end sip.conf - Att, Anderson Luiz Brunozi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Tuesday, July 29, 2008 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk SIP configuration Anderson Luiz Brunozi wrote: Hello, I'm trying to setup an instance of Asterisk to use as a SIP conference server only. And I'm using OpenSER as SIP Registrar/Proxy. On asterisk's sip.conf I created a user [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (sipdomain == server name), that registers to OpenSER when asterisk is started. The user conference refers to context conference-context. And, on the extensions.conf I've defined the context called conference-context, with an extension named conference. From my SIP client, I call sip:[EMAIL PROTECTED] And get a 404 Not found response from asterisk. On asterisk's log I see messages like: Looking for conference on conference-context (domain serverIP) And: Call from 'conference' to extension 'conference' rejected because extension not found. Does anyone have an ideia of why I'm getting that message? Why does asterisk seem to be using domain == serverIP, instead of domain == servername? Is that correct the behavior? Or I may have something missing on my configuration? I suspect the problem is more likely be in the sip.conf file, but I can't see what's wrong/missing. I'm using app_conference. But I don't think this matters for now, because my first line of the conference extension calls Log(). And, as I don't see my log message printed, I assume asterisk didn't even start processing the commands defined for the conference extension. Thanks, Anderson Luiz Brunozi -- -- ___ -- 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 Please send the relevant portion of your extensions.conf, as that is where the problem is -- Sherwood McGowan VoIP / Telecom Solutions [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] Addressbook solution for Cisco 7961?
Hi, I'm looking for an addressbook solution that works with Cisco 7961 (SIP 8.3.5 firmware) so it's available as a service by pressing the button with the picture of the globe on it. Suggestions most welcome. Regards, Patrick ___ -- 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] Addressbook solution for Cisco 7961?
Read my hacks on the Cisco phones in Oreilly's VoIP Hacks book On Tue, Jul 29, 2008 at 12:26 PM, Patrick [EMAIL PROTECTED] wrote: Hi, I'm looking for an addressbook solution that works with Cisco 7961 (SIP 8.3.5 firmware) so it's available as a service by pressing the button with the picture of the globe on it. Suggestions most welcome. Regards, Patrick ___ -- 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 -- Andrew lathama Latham Principal TuxTone Inc. http://TuxTone.com [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
[asterisk-users] Purchasing Digium IVR Prompts.
Just went to order some IVR prompts from the digium web site From the digium web site: We have created an easy and cost effective way to have customized recordings done quickly and with no hassle. I thought this was rather amusing, as: 1. If you want multiple prompts recorded, you need to submit a new order for each, which means that even prompts of a couple of words are still charged at $12. That is NOT cost effective. You could record all your prompts as a single order, but then you'd need to split up the prompts yourself with audio software. That is NOT hassle free. 2. Since prompts are recorded seperately, each shows up in the shopping cart as a separate item. There is no way to see what the requested prompt is! We're going to have a lot of these (remember, each prompt is different), and keeping track of them NOT hassle free. 3. From the web site Also, you have the ability to upload your own intonation file to ensure a personalized and professional recording every time. what the heck is an intonation file? Is it a text file? Is it an audio recording? What format? The web site doesn't seem to say. Lack of documentation on the web site is NOT hassle free. 4. Of course, when I called customer service, they had no clue. NOT hassle free. Doug. ___ -- 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] Purchasing Digium IVR Prompts.
Try GM Voices. $6.95 per prompt plus $175 studio setup fee. To make it truly cost effective it might be worth it to find other users who need prompts recorded and then you can split the setup fee. Even if you have dozens or hundreds of prompts the fee is what is. I think they charge a separate fee for each voice talent so if you need prompts in different languages you'll have a setup fee for each language. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Tuesday, July 29, 2008 10:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Purchasing Digium IVR Prompts. Just went to order some IVR prompts from the digium web site From the digium web site: We have created an easy and cost effective way to have customized recordings done quickly and with no hassle. I thought this was rather amusing, as: 1. If you want multiple prompts recorded, you need to submit a new order for each, which means that even prompts of a couple of words are still charged at $12. That is NOT cost effective. You could record all your prompts as a single order, but then you'd need to split up the prompts yourself with audio software. That is NOT hassle free. 2. Since prompts are recorded seperately, each shows up in the shopping cart as a separate item. There is no way to see what the requested prompt is! We're going to have a lot of these (remember, each prompt is different), and keeping track of them NOT hassle free. 3. From the web site Also, you have the ability to upload your own intonation file to ensure a personalized and professional recording every time. what the heck is an intonation file? Is it a text file? Is it an audio recording? What format? The web site doesn't seem to say. Lack of documentation on the web site is NOT hassle free. 4. Of course, when I called customer service, they had no clue. NOT hassle free. Doug. ___ -- 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] soundpoint 301 power adapter output?
Can anybody confirm if this is the correct power adapter outputs: 12V DC 400mA You adapter will have to outputs listed on it. Thanks, PB ___ -- 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] Auto Dial Application
Hello List, Looking for an Automatic Dial app that can play announcements as soon as the phone is answered. Please contact offlist. K Singh http://www.couponcactus.com/?kpm=1 ___ -- 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] Purchasing Digium IVR Prompts.
I've used http://www.pbxprompts.com/ The whole pack is around 100$ and then I think I was charged 11$ per prompt for custom ones. No setup fee that I recall. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Tuesday, July 29, 2008 1:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Purchasing Digium IVR Prompts. Try GM Voices. $6.95 per prompt plus $175 studio setup fee. To make it truly cost effective it might be worth it to find other users who need prompts recorded and then you can split the setup fee. Even if you have dozens or hundreds of prompts the fee is what is. I think they charge a separate fee for each voice talent so if you need prompts in different languages you'll have a setup fee for each language. -MC _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Tuesday, July 29, 2008 10:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Purchasing Digium IVR Prompts. Just went to order some IVR prompts from the digium web site From the digium web site: We have created an easy and cost effective way to have customized recordings done quickly and with no hassle. I thought this was rather amusing, as: 1. If you want multiple prompts recorded, you need to submit a new order for each, which means that even prompts of a couple of words are still charged at $12. That is NOT cost effective. You could record all your prompts as a single order, but then you'd need to split up the prompts yourself with audio software. That is NOT hassle free. 2. Since prompts are recorded seperately, each shows up in the shopping cart as a separate item. There is no way to see what the requested prompt is! We're going to have a lot of these (remember, each prompt is different), and keeping track of them NOT hassle free. 3. From the web site Also, you have the ability to upload your own intonation file to ensure a personalized and professional recording every time. what the heck is an intonation file? Is it a text file? Is it an audio recording? What format? The web site doesn't seem to say. Lack of documentation on the web site is NOT hassle free. 4. Of course, when I called customer service, they had no clue. NOT hassle free. Doug. ___ -- 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] Recommend Bluetooth adapters for chan_mobile?
I've been experimenting with chan_mobile, and am finding it's less than reliable using the bargain-basement class 2 USB adapter I had lying around. I've seen good reports in various posts around the 'net, and suspect if I were to get a better USB adapter (two, actually - two phones) then I'd see better results. I think it'll need to be class 1 (100m range) rather than class 2 (10m range) because of the layout of my home (it's not big, just a few walls in the way). Googling didn't turn up a lot of useful information, except a common recommendation to use CSR-based adapters - but little (current) info on which adapters are (my current one isn't). Can anyone share their experiences and specific recommendations for Bluetooth USB adapters to use with chan_mobile? (available in the United States). Any other sage advice for getting the most from chan_mobile? Thanks in advance, Paul (Fedora 9, with bluez 3.32 and asterisk 1.6.0-0.17.beta9 versions of asterisk and asterisk-mobile RPMs installed) ___ -- 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] soundpoint 301 power adapter output?
Paul Belanger wrote: Can anybody confirm if this is the correct power adapter outputs: 12V DC 400mA You adapter will have to outputs listed on it. Thanks, PB That is correct for a 301. -Dave ___ -- 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] Callerid Woes
I am trying to setup one time caller id block on my system(activated
when an incoming call matches *811XX), and I have had little to
no luck. Could you take a look at my context/macro definition and help
me figure out what I am missing?
Here is my context for my dialplan:
include=default
plancomment=user-default
exten=_1XX!,1,Macro(trunkdial,${my_trunk}/${EXTEN:0},${trunk_cid})
comment=_1XX!,1,my_trunk,standard
exten=_*811XX!,1,Macro(private_trunkdial,${my_trunk}/${EXTEN:3},null)
comment=_*811XX!,1,my_trunk,standard
(I replaced the name of my termination wih my_trunk)
And, below is the the private_trunkdial macro that is used above:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
exten=s,n,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Hangup
exten=s-BUSY,1,Hangup
exten=_s-.,1,NoOp
I know something is off, because according to terminating VoIP provider,
nothing is actually removed from my call request, and all my
information remains unaltered. Any guidance would be appreciated.
John Koenig
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[asterisk-users] Fallback on a fallback
I have two sites running Asterisk PBX. Normally the inbound calls go
through a 3rd (colocated) server and are routed via IAX to the site
(the site registers with the main server)
I created a macro that tries to ring one location and then another.
Each site explicitly Answer() the call even though it will only ring
all the sip phones at the relevant location. When fall back is in
effect it goes to the other location and then the other PBX server
rings the same phones (they register to both servers, the server at
the other location via VPN). It was implemented this way because at
the time there was a hardware stability issue.
Now I want to add a 3rd failback via a PSTN line. This will be done
from the main colocated server so even if the internet at the
location is down calls go to the PBX via the PSTN and if the PBX
server catches fire we setup some Weco 2500 clones (in red) as further
protection. But the issue here is that if we tell it to ring (in this
order) site 1, site 2 and then PSTN and the internet to site 1 is down
it will go to site 2 and be answered but since the internet is down
so is the VPN and the call drops there. I can change that, but if only
the PBX server is down (and not the internet or VPN) then I don't want
to use the PSTN line because capacity is only 1 call inbound or
outbound and any subsequent callers would get a busy tone. I also
don't want to send the call out of site 2 directly due to bandwidth
concerns.
Does anyone have a suggestion on how to implement this?
Current setup is exactly as follows
MAIN:
;exten = 13057221371,1,Macro(welcome-message)
;exten = 13057221371,n,Macro(site-fallback,site1/4997|site2/4997|7|7)
[macro-site-fallback]
; ${ARG1) Dialstring 1
; ${ARG2} Dialstring 2
; ${ARG3} Ringtime Peer 1
; ${ARG4} Ringtime PEER 2
exten = s,1,Playtones(ring)
exten = s,2,Dial(${ARG1},${ARG3},m)
exten = s,n,Goto(s-${DIALSTATUS},1)
;exten = s-NOANSWER,1,
;exten = s-BUSY,1,Macro(all-circuits-busy)
;exten = s-BUSY,n,Hangup
exten = _s-.,1,GoTo(s-BACKUP,1)
exten = s-BACKUP,1,Dial(${ARG2},${ARG4},m)
exten = s-BACKUP,n,Goto(s-BACKUP-${DIALSTATUS},1)
exten = s-BACKUP-NOANSWER,1,Macro(no-answer)
exten = s-BACKUP-NOANSWER,n,Hangup
exten = s-BACKUP-BUSY,1,Macro(all-circuits-busy)
exten = s-BACKUP-BUSY,n,Hangup
exten = _s-BACKUP.,1,Macro(network-error)
exten = _s-BACKUP.,n,Hangup
Site 1 or 2 (they are basically identical) but FWIW this is the config
of site 2 for failover of site 1:
exten = 4997,1,Answer
exten = 4997,n,Set(CALLERID(name)=CM Fallback Service})
exten =
4997,n,Dial(SIP/401SIP/402SIP/403SIP/404SIP/405SIP/406SIP/407SIP/408SIP/409SIP/410,90,r)
exten = 4997,n,Playtones(ring)
exten = 4997,n,Wait(1)
exten = 4997,n,VoiceMail(499|u)
pbxserver-sitetwo*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
410/410192.168.12.111 D 5060 Unmonitored
409/409192.168.12.100 D 5060 Unmonitored
408/408192.168.12.116 D 5060 Unmonitored
406/406(Unspecified)D 0Unmonitored
405/405192.168.12.223 D 5060 Unmonitored
404/404(Unspecified)D 0Unmonitored
403/403192.168.12.248 D 5060 Unmonitored
402/402(Unspecified)D 0Unmonitored
401/401192.168.12.119 D 5060 Unmonitored
210/210192.168.0.253D 5060 OK (39 ms)
209/209192.168.0.106D 5060 OK (40 ms)
208/208192.168.0.190D 5060 OK (40 ms)
207(Unspecified)D 0UNKNOWN
206/206192.168.0.194D 5060 OK (38 ms)
205/205192.168.0.105D 5060 OK (43 ms)
204/204192.168.0.173D 5060 OK (39 ms)
203/203192.168.0.126D 5060 OK (37 ms)
202/202192.168.0.187D 5060 OK (39 ms)
201/201192.168.0.176D 5060 OK (40 ms)
501/501(Unspecified)D 0UNKNOWN
20 sip peers [18 online , 2 offline]
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Re: [asterisk-users] Callerid Woes
John Koenig wrote:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
Just a guess.
exten = s,1,Set(CALLERID(all)= null 0)
exten = s,n,SetCallerPres(prohib)
exten = s,n,Dial(${ARG1})
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
China for E1 as has been mentioned is a strange place. China is broken up into two parts basically south of Shanghai and north of Shanghai. It is all operated by China Telecom and CNC, but the technical aspects of the two parts are not necessarily the same. For technical assistance in China we would be very please to assist as the Digium distributor for the region with offices in Beijing, Hong Kong and Sydney. Our technical support people would be very pleased to assist any companies or individuals off line. Email address below. Rupert Utteridge Director - Sales Marketing Digital Techniques (Beijing) Limited Room 0209, Tower 2 Beijing Bright China Chang An Building 7 Jianguomen Nei Avenue Beijing 15 People's Republic of China Tel: +86 10 6510 1588 Fax: +86 10 6510 1587 Email: [EMAIL PROTECTED] Web: www.dtasia.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callerid Woes
Doug Lytle wrote:
John Koenig wrote:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
Just a guess.
exten = s,1,Set(CALLERID(all)= null 0)
exten = s,n,SetCallerPres(prohib)
exten = s,n,Dial(${ARG1})
Doug
I believe you need to use:
exten = s,1,Set(CALLERID(all)=)
To set an empty callerId
--
JohnM
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Re: [asterisk-users] Realtime + SIP + MySQL: md5secret BROKEN
mysql alter table sip_buddies drop md5secret;
Query OK, 1 row affected (0.00 sec)
Records: 1 Duplicates: 0 Warnings: 0
Suddenly, authentication works!
The md5secret used was the md5 of 'qwedsa', and the value was correct.
mysql select md5('qwedsa');
+--+
| md5('qwedsa')|
+--+
| 4d27b7677bd96f7ba00c4bd0541c9588 |
+--+
1 row in set (0.00 sec)
Walter,
Not sure, but the above might be your problem.
The md5secret is NOT a MD5 sum of the secret, but of the combination
username:realm:secret. So in your case you should add this md5secret:
mysql select md5('walter:asterisk:qwedsa');
+--+
| md5('walter:asterisk:qwedsa')|
+--+
| 577061918968e961153393ef87b43e4b |
+--+
This would explain why the tests with cleartext secrets work, and not the
ones with the md5secret. Not sure if you tried md5secrets with a static
sip.conf user definition, but the result should be a credential failure in
that case as well.
Best regards,
Torbjörn
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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Lee, John (Sydney) wrote: This time, I am trying to remotely install Asterisk in China. I was told that an E1 line has been installed and so I plug it into port 1 of a TE412P. i've installed several Asterisk systems in Shanghai Beijing. On the box, first of all, I just installed Zaptel 1.4.10.1. # service zaptel restart Unloading zaptel hardware drivers:ERROR: Module zaptel is in use . Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. Running ztcfg: [ OK ] you might want to comment out all other modules in /etc/default/zaptel except for wct4xxp (if that's the only zaptel card you have). # vi zaptel.conf [...] span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 this looks right. however crc4 is optional. you have to check with the phone company. sometime they do require it other time they don't. it's not very consistent. *** However, I received a red alarm in zttool and the LED on the TE412P card is also red. *** I have made sure that the jumper is closed for port 1 on the TE412P card and so it could not be the jumper problem. red alarm usually means there's no clocking signal. check all your cables (crossover vs straight through) if the cable's good. call phone company and complain. in my experience 9 out of 10 time we have to call phone company and complain. ### Because this is the first time I install Asterisk in China and I was wondering if their E1 is different from the Euro E1. ### However, I went into dmesg and I discovered the following. ### Could it really be a zaptel bug? I saw on a similar few on the digium bug list but I cannot be 100% sure. Any thoughts? About to enter spanconfig! Done with spanconfig! Registered tone zone 33 (China) About to enter startup! TE4XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 128 channels BUG: soft lockup - CPU#2 stuck for 16s! [ztcfg:4681] Pid: 4681, comm:ztcfg EIP: 0060:[f8cba1df] CPU: 2 EIP is at init_vpm450m+0x32d/0x34a [wct4xxp] EFLAGS: 0286Tainted: G (2.6.18-92.1.6.el5 #1) EAX: EBX: f76ae8f0 ECX: 0019 EDX: ESI: f7997400 EDI: 0286 EBP: f76838c0 DS: 007b ES: 007b CR0: 8005003b CR2: b7f01007 CR3: 37bc1000 CR4: 06d0 [f8ca1b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8ca5ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042609c] release_console_sem+0x17e/0x1b8 [c046d53a] cache_alloc_refill+0x14b/0x450 [f8956f61] zt_ioctl+0x273/0x144f [zaptel] [c04d7d45] generic_make_request+0x248/0x258 [c045ae3c] __do_page_cache_readahead+0x69/0x1c6 [c0484a5b] __d_lookup+0x98/0xdb [c047c110] do_lookup+0x53/0x166 [c047e7e4] do_path_lookup+0x20e/0x25e [c047c389] permission+0xa2/0xb5 [c04e2d06] kobject_get+0xf/0x13 [c046f7fa] __dentry_open+0xea/0x1ab [c046f91f] nameidata_to_filp+0x19/0x28 [c046f959] do_filp_open+0x2b/0x31 [c048029b] do_ioctl+0x47/0x5d [c04804fb] vfs_ioctl+0x24a/0x25c [c0471bbe] __fput+0x13f/0x167 [c0480555] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) Completed startup! i've seen that before. (forgot which version of zaptel). it went away after i upgraded it. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- 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] Callerid Woes
I tried all of the suggestions, and still the callerid remains intact.
I guess at this point I am starting to wonder what bit of logic is being
run when I dial *8111XX...
Is there a way I can trace how a call is being processed within
asterisk? Or even see what I am sending to my VoIP terminating node?
John
John Millican wrote:
Doug Lytle wrote:
John Koenig wrote:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
Just a guess.
exten = s,1,Set(CALLERID(all)= null 0)
exten = s,n,SetCallerPres(prohib)
exten = s,n,Dial(${ARG1})
Doug
I believe you need to use:
exten = s,1,Set(CALLERID(all)=)
To set an empty callerId
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Re: [asterisk-users] Callerid Woes
John Koenig wrote:
I tried all of the suggestions, and still the callerid remains intact.
I guess at this point I am starting to wonder what bit of logic is being
run when I dial *8111XX...
Is there a way I can trace how a call is being processed within
asterisk? Or even see what I am sending to my VoIP terminating node?
John
John Millican wrote:
Doug Lytle wrote:
John Koenig wrote:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
Just a guess.
exten = s,1,Set(CALLERID(all)= null 0)
exten = s,n,SetCallerPres(prohib)
exten = s,n,Dial(${ARG1})
Doug
I believe you need to use:
exten = s,1,Set(CALLERID(all)=)
To set an empty callerId
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Yes, install wireshark and run tethereal port 5060 -w capture.pcap and
then place a call...then open the pcap file in Wireshark and look in the
SIP header for the callerid information
--
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED]
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Re: [asterisk-users] Callerid Woes
John Koenig wrote:
I tried all of the suggestions, and still the callerid remains intact.
I guess at this point I am starting to wonder what bit of logic is being
run when I dial *8111XX...
Is there a way I can trace how a call is being processed within
asterisk? Or even see what I am sending to my VoIP terminating node?
John
John Millican wrote:
Doug Lytle wrote:
John Koenig wrote:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
Just a guess.
exten = s,1,Set(CALLERID(all)= null 0)
exten = s,n,SetCallerPres(prohib)
exten = s,n,Dial(${ARG1})
Doug
I believe you need to use:
exten = s,1,Set(CALLERID(all)=)
To set an empty callerId
typing:
sip set debug peer peer_name
at the CLI will give you a bunch of information as to what is going on
with that peer
--
JohnM
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Re: [asterisk-users] Callerid Woes
Thanks for the tip about sip set debug peer. I was able to capture
some information about the call in progress, but I am confused as to
what I see. When I pick up my sip phone I dial
*811area_codeprefixnumber, and the first invite I see is going to
1area_codeprefixnumber@my_asterisk_ip. Shouldn't the *81 be
included in the request?
Is it possible that the linksys pap2 that I am using is removing the *81
prior to placing the invite request?
John
John Millican wrote:
John Koenig wrote:
I tried all of the suggestions, and still the callerid remains intact.
I guess at this point I am starting to wonder what bit of logic is being
run when I dial *8111XX...
Is there a way I can trace how a call is being processed within
asterisk? Or even see what I am sending to my VoIP terminating node?
John
John Millican wrote:
Doug Lytle wrote:
John Koenig wrote:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
Just a guess.
exten = s,1,Set(CALLERID(all)= null 0)
exten = s,n,SetCallerPres(prohib)
exten = s,n,Dial(${ARG1})
Doug
I believe you need to use:
exten = s,1,Set(CALLERID(all)=)
To set an empty callerId
typing:
sip set debug peer peer_name
at the CLI will give you a bunch of information as to what is going on
with that peer
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Re: [asterisk-users] Callerid Woes
John Koenig wrote:
Thanks for the tip about sip set debug peer. I was able to capture
some information about the call in progress, but I am confused as to
what I see. When I pick up my sip phone I dial
*811area_codeprefixnumber, and the first invite I see is going to
1area_codeprefixnumber@my_asterisk_ip. Shouldn't the *81 be
included in the request?
Is it possible that the linksys pap2 that I am using is removing the *81
prior to placing the invite request?
John
John Millican wrote:
John Koenig wrote:
I tried all of the suggestions, and still the callerid remains intact.
I guess at this point I am starting to wonder what bit of logic is being
run when I dial *8111XX...
Is there a way I can trace how a call is being processed within
asterisk? Or even see what I am sending to my VoIP terminating node?
John
John Millican wrote:
Doug Lytle wrote:
John Koenig wrote:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
Just a guess.
exten = s,1,Set(CALLERID(all)= null 0)
exten = s,n,SetCallerPres(prohib)
exten = s,n,Dial(${ARG1})
Doug
I believe you need to use:
exten = s,1,Set(CALLERID(all)=)
To set an empty callerId
typing:
sip set debug peer peer_name
at the CLI will give you a bunch of information as to what is going on
with that peer
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The Linksys is taking *81 as a local spree code, causing it to be
stripped. The problem you're then getting is probably that Asterisk is
using _it's_ caller id information for your peer
--
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED]
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[asterisk-users] Multiple Asterisk SIP Server/client connections
I have 4 asterisk servers. They all have local phones on their local network they manage for SIP based conversations. We then have IAX between them all for inter-asterisk connections. This setup has worked well for nearly 2 years now, minor problems here and there but overall very nice. Recently we acquired some Polycom video conference units. I was able to setup our main server to host all the video coordination using video over SIP. I was able to configure the video conference units on the local network, have all 4 of them (one going to each remote server) displaying 4 videos on the local network. I then sent them out to their remote facilities and setup Asterisk with as a SIP client on the 3 remote locations to talk to the main server. One at a time we tested them and they worked one on one. Recently we tried to get two going, and I noticed there seems to be an issue with the SIP registration if one of the 3 remote SIP clients has already registered. That is, the other requests are unanswered or not fully registered for some reason or another. At very random times I've actually managed to get 2 of the 3 connected, but inevitably I lose one of those 2 shortly after. The SIP.CONF has been made identical across all 3 remote locations, and the main server has the same config for each remote site connecting. I first want to confirm that it's possible to have 3 remote Asterisk servers setup as a SIP client connected to a 4th Asterisk server. Assuming it is possible, here is the SIP Client SIP.CONF: [general] register = 103:[EMAIL PROTECTED]/699 defaultexpirey=1800 maxexpirey=3600 relaxdtmf=yes videosupport=yes disallow=all allow=ulaw allow=gsm allow=h263p canreinvite=no limitonpeer=yes notifyringing=yes notifyhold=yes externip=xx.xx.xx.xx.xx fromdomain=xx.xx.xx.xx localnet=192.168.0.0/255.255.255.0 [yy.yy.yy.yy] type=friend host=yy.yy.yy.yy insecure=port,invite [699] type=friend secret=1234 dial=SIP/699 callerid=Video 699 allowsubscribe=yes host=dynamic context=from-internal insecure=port,invite In addition here's the relevant portions of the SIP.CONF from the main server: [general] videosupport=yes disallow=all allow=ulaw allow=gsm allow=h263p canreinvite=no fromdomain=yy.yy.yy.yy externip=yy.yy.yy.yy localnet=10.200.26.0/255.255.255.0 nat=yes bindport=5060 [103] type=friend secret=1234 dial=SIP/103 callerid=Video103 allowsubscribe=yes host=dynamic context=from-internal insecure=port,invite Please, any suggestions would be great. I've been bashing my head against the keyboard all day trying to find why it's acting in this way. ___ -- 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+ fax-to-mail
Nadjia Boumédiène wrote: Hello, I have asterisk 1.2.7 and I would like to install the fax-to-mail. I already installed spandsp (2.25), app_rxfax, app_txfax and app_makefile.patch , I rebuilt Asterisk but when I send a fax, I don’t receive it. I have this: Without knowing the contents of mailfax, I'd have to guess that you aren't running a mail server (Postfix/Sendmail) on the system handing the fax2email. 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
[asterisk-users] mpg123
Hey, I recently deployed a box with a dialplan that needs to play mp3's. I had mpg123 on it and I'm using MP3Player to play the mp3s. However when I test now instead of the mp3's being played I get a piano like melody. It goes like bing bing bing, don don don, etc like someone doing repetition of keys on a piano. Anyone ever experience this? Regards, Igor H. ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
An output of 'head -n 1 /proc/zaptel/*' after the fact might help us help you. Did you re-run ztcfg after editing zaptel.conf ? On a sidebar, let me suggest head -1q; it's neater. Thanks Jay for your neater suggestion! # head -n1 /proc/zaptel/* == /proc/zaptel/1 == Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED == /proc/zaptel/2 == Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 == /proc/zaptel/3 == Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 == /proc/zaptel/4 == Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # head -1q /proc/zaptel/* Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
Thanks Steve for your suggestions. In China you will generally get either MFC/R2 or EuroISDN. MFC/R2 is much more common. This is exactly my current problem. NETCOM in Shanghai just told my local contact it is an E1 and that's it. I have no idea whether it is MFC/R2 or EuroISDN and so there is a lot of trial and error, not to mention about communicating with the telco. Is there anyway I could find out from zaptel what the line signal is? The only oddity with EuroISDN is that it often provided without CRC4. That doesn't make a lot of sense, but there it is. MFC/R2 seems to be universally provided without CRC4 in China. That's great info, Steve. You said you are sure the card is OK. How did you determine that? Have you tried a loopback cable between two its its ports? Are you sure the cable you have used is OK? I am quite sure that the Digium card is fine because it is a new card from Digium and I sent it from here. My experience with Digium cards has been good. As far as the cable goes, this is a bit complicated. The way it works is the telco delivers a fibre optic cable to the floor and the fibre terminates on a fibre optic multiplexer. Then the multiplexer is connected to a Fast Ethernet to E1 converter which has a RJ45 port. We then connect this RJ45 port to the TE412P port. Anyway, the quality of the data cables in China appears to be really bad and I will have to send some better quality cables from down under to them to try. ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
If you're interested, our working circuit in Chengdu with a China Telecom PRI is configured as: /etc/zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = cn defaultzone=cn /etc/asterisk/zapata.conf (just including the pertinent lines): switchtype=euroisdn signalling=pri_cpe group = 1 channel=1-15,17-31 Thanks Chris. I have tried exactly the same zaptel.conf before but I got red alarm on the card and zttool. I have not configured zapata.conf yet because I wanted to get zaptel right first. The installation I am working on is in Shanghai and the line is provided by NETCOM. ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
i've installed several Asterisk systems in Shanghai Beijing. Thanks Edwin. The remote site is in Shanghai and NETCOM is the telco. Do you know if their E1 line is MFC/R2 or EuroISDN? red alarm usually means there's no clocking signal. check all your cables (crossover vs straight through) As far as the cable goes, this is a bit complicated. The way it works is the telco delivers a fibre optic cable to the floor and the fibre terminates on a fibre optic multiplexer. Then the multiplexer is connected to a Fast Ethernet to E1 converter which has a RJ45 port. We then connect this RJ45 port to the TE412P port. Anyway what you said is still a good point - I will try replacing the straight through cable with a crossover and give it a go. if the cable's good. call phone company and complain. in my experience 9 out of 10 time we have to call phone company and complain. How should we complain? Are there any technical details we need to show them? It is a different country though. ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
Yes, to try port 4 again. Thanks Tzafrir. Why do I have to plug it into port 4? Backup the existing zaptel.conf and run genzaptelconf (no need to unload / reload any modules). What is the output of 'head -n 1 /proc/zaptel/*' after that? I could not find genzaptelconf probably because I deselect menuselect xpp when I installed zaptel because of that xpp compile bug. I have to find the workaround to get xpp compiled then. ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
On Wed, Jul 30, 2008 at 01:38:11PM +1000, Lee, John (Sydney) wrote: Yes, to try port 4 again. Thanks Tzafrir. Why do I have to plug it into port 4? Backup the existing zaptel.conf and run genzaptelconf (no need to unload / reload any modules). What is the output of 'head -n 1 /proc/zaptel/*' after that? I could not find genzaptelconf probably because I deselect menuselect xpp when I installed zaptel because of that xpp compile bug. I have to find the workaround to get xpp compiled then. You don't need to install it. Just run kernel/xpp/utils/genzaptelconf directly from the source directory. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
You don't need to install it. Just run kernel/xpp/utils/genzaptelconf directly from the source directory. Yes mate - I was just 1 sec away from reinstalling zaptel. Why do I have to plug it into port 4? Do I have to plug the line into port 4 instead of port 1? ___ -- 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] Newbie in China: Red alaram in Zaptel for E1
John wrote: Thanks Steve for your suggestions. In China you will generally get either MFC/R2 or EuroISDN. MFC/R2 is much more common. This is exactly my current problem. NETCOM in Shanghai just told my local contact it is an E1 and that's it. I have no idea whether it is MFC/R2 or EuroISDN and so there is a lot of trial and error, not to mention about communicating with the telco. Is there anyway I could find out from zaptel what the line signal is? International installs are always fun. I have had some luck getting a local employee to relay my questions about provisioning, but all to often the response is 'We use the standard settings...'. At that point I resort to trial and error. I have setup a circuit in Shanghai, it is an E1, CRC4/HDB3 with the telco switch being/or compatible with ATT 5ESS. You should be able to get Netcom to tell you if the circuit is ISDN or not. Asking if it is a PRI will just confuse them, but they do understand the question 'ISDN or not ISDN' The only oddity with EuroISDN is that it often provided without CRC4. That doesn't make a lot of sense, but there it is. MFC/R2 seems to be universally provided without CRC4 in China. That's great info, Steve. Dan ___ -- 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
