Re: [Asterisk-Users] real-time priority , -p switch
On Thu, 11 Aug 2005, Joseph wrote: In this case could somebody explain to me why run asterisk with ''-p switch? According to asterisk man explanation for -p is as follow: If supported by the operating system (and executing as root), attempt to run with realtime priority for increased performance and responsiveness within the Asterisk process, at the expense of other programs running on the same machine. Since Linux is not RTOS, why some folks are using this -p switch? It has no effect on standard Linux box. Linux is not a hard realtime os with guaranteed timing. What the -p flag does is to request the realtime scheduler. This means a process wil no longer be subjected to the stanadrd unix scheduling but rather use a strict priority scheduling. The net result is that once a process using the realtime scheduler is ready to run the kernel wihh schedule it as soon as possible. It will only be preempted by realtime processes of the same or better priority. With the addition of the lowlatency patches the worst case latency for userspace applications is very low. The remaining difference between a hard RT os is the guarantees it can make. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] list in asterisk cli is getting too long
- Original Message - From: Ronald Wiplinger To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, August 12, 2005 6:04 AM Subject: [Asterisk-Users] list in asterisk cli is getting too long How can I use something like|morein CLI ? The lists are getting too long, like sip show users Hi If all you need is to browse the list of users, and you don't need to be in the CLI all the time, you can do something like: asterisk -rx sip show users | more from the Unix command line. Regards Hilton Datatex Dynamics CC Web site http://www.datatex.co.za/ Email to [EMAIL PROTECTED] Tel +27215924033 Fax +27215924077 The use of the Datatex e-mail facility is not permitted for the distribution of chain letters or offensive email of any nature whatsoever. Datatex hereby distances itself from and accepts no liability in respect of the unauthorised use of its e-mail facility or the sending of e-mail communications for other than strictly business purposes. Datatex furthermore disclaims liability for any unauthorised instruction for which permission was not granted. Any recipient of an unacceptable communication, a chain letter or offensive material of any nature is requested to report it to [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
On Thu, 11 Aug 2005, Geoff Manning wrote: We are having line noise issues in our Asterisk to legacy PBX integration. All SIP calls originating from IP phones sound crystal clear. All calls that originate from the legacy PBX (Isoetec 228) and route through the Asterisk and out SIP have a lot of line noise. I believe I have it pinned down to these Blue Alarm errors that I can see on the legacy PBX side. zttool shows no alarm but when I view the T1 stats on the Isoetec I see numerous Blue Alarms. A blue alarm sounds really strange. That indicates that the remote end (asterisk) in this case does not want to play at all. On a T1 it is sent as a continous series of unframed 1:s. I am not sure if asterisk ever sends a blue alarm (Alarm Indication Signal). Receiving a blue alarm is indicative of a serious problem. There should not be any audio at that time, since the blue alarm is actually a long unframed signal. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79XX and VLANS
If you're using Cisco Switches: Logon to the switch and go to config mode int fa0/1 switchport access voice vlan untagged sometheing in that direction configures the CDP to set the phone to untagged frames. Julian Lyndon-Smith wrote: How ? Where ? I've been wanting to do this for ages, and never found an option to do so ! Please Please Please tell all. (I hate begging, but sometimes ) Julian. Eric Wieling aka ManxPower wrote: Matthew Boehm wrote: Hey gang, We have about 30 Cisco 79XX phones all running latest 7.5 SIP. We are also using all Cisco Switches and Routers. Everything works great except that when you reboot a phone it takes like 3-5 minutes for it to come up. The phones spend tons of time 'Configuring VLAN..' We don't run any VLANs. Is there some way to skip this? In the 'Network Settings' I have both 'Operational VLAN Id' and 'Admin VLAN Id' set to blank values. Disable CDP on the phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE405P V2 changes?
Hi I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after putting in the upgraded board but did not change any conf, but the spans become active but will not come up. I guess I am missing something or are the any changes to the zaptel/libpri software that is required. I cannot find any info about this or does this new firmware only work with latest CVS. I am using 1.0.9 with 2.6.12 kernel Zapata Telephony Interface Registered on major 196 Found TE4XXP at base address fdfff000, remapped to f8928000 TE4XXP version c01a0164, burst ON, slip debug: OFF TE4XXP running with work queues. FALC version: 0005, Board ID: 00 Reg 0: 0x364e9400 Reg 1: 0x364e9000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a0164 Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE405P (2nd Gen) eth0: link up, 10Mbps, half-duplex, lpa 0x About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Unassigning channel 0/1! Unassigning channel 0/2! Unassigning channel 0/3! Unassigning channel 0/4! Unassigning channel 0/5! Unassigning channel 0/6! Unassigning channel 0/7! Unassigning channel 0/8! Unassigning channel 0/9! Unassigning channel 0/10! Unassigning channel 0/11! Unassigning channel 0/12! Unassigning channel 0/13! Unassigning channel 0/14! Unassigning channel 0/15! Unassigning channel 0/16! Unassigning channel 0/17! Unassigning channel 0/18! etc... This was working for 10 months before the upgrade. Master ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] My users are using PSTN instead of VoIP
Some of my customers are using PSTN to call each other from the VoIP system. I want to stop that, by setting up all internal numbers to be reachable via VoIP first. E.g. A calls B via VoipJet !!! But B is on our system. I want now set it up so that if B is reachable via VoIP, than it should call this number directly. I could now setup all these extensions and avoid so VoipJet, but I wonder, if not there is a simpler way via private ENUM .. Has anybody done that? bye Ronald Wiplinger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] txgain for SIP?
Is there an option txgain for SIP in Asterisk? My users all complain that their other parties think that they are way too silent even though they all have their mic volume all the way up and also enabled the 'mic boost' option. This happens with all the clients that we're using and also with different model headsets, so my last hope is txgain for SIP. Thanks Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZapHFC E1 PRI (cwain)
Hello, I've got a Junghanns ZapHFC E1 PRI Card (cwain) and this driver writes very much messages into /var/log/messages like the following: --- snip --- Aug 2 17:58:02 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x37 0x90 0xc3 ] 6 bytes Aug 2 17:58:02 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xef ] Aug 2 17:58:02 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:12 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x37 0x90 0xc3 ] 6 bytes Aug 2 17:58:12 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xef ] Aug 2 17:58:12 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:22 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x37 0x90 0xc3 ] 6 bytes Aug 2 17:58:22 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xef ] Aug 2 17:58:22 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:23 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x37 0x90 0xc3 ] 6 bytes Aug 2 17:58:23 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xef ] Aug 2 17:58:23 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:28 asterisk1 kernel: cwain: card 1 TX [ 0x0 0x1 0x36 0xee 0x8 0x2 0x0 0x68 0x5 0x4 0x3 0x80 0x90 0xa3 0x18 0x3 0xa9 0x83 0x81 0x6c 0xc 0x41 0x81 0x32 0x31 0x33 0x31 0x36 0x36 0x35 0x31 0x33 0x34 0x70 0xc 0xc1 0x36 0x34 0x34 0x31 0x31 0x37 0x31 0x39 0x31 0x39 0x30 0xa1 ] Aug 2 17:58:28 asterisk1 kernel: ztx 48 bytes Aug 2 17:58:28 asterisk1 kernel: cwain: card 1 RX [ 0x0 0x1 0x1 0x38 0x11 0x2 ] 6 bytes Aug 2 17:58:28 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0xee 0x38 0x8 0x2 0x80 0x68 0x2 0x18 0x3 0xa9 0x83 0x81 0x1e 0x2 0x82 0x88 0x51 0x20 ] 20 bytes Aug 2 17:58:28 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf0 ] Aug 2 17:58:28 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:28 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0xf0 0x38 0x8 0x2 0x80 0x68 0x1 0xdd 0xf ] 11 bytes Aug 2 17:58:28 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf2 ] Aug 2 17:58:28 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:31 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0xf2 0x38 0x8 0x2 0x80 0x68 0x45 0x8 0x2 0x84 0x91 0x1e 0x2 0x82 0x88 0xdc 0xc9 ] 19 bytes Aug 2 17:58:31 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf4 ] Aug 2 17:58:31 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:31 asterisk1 kernel: cwain: card 1 TX [ 0x0 0x1 0x38 0xf4 0x8 0x2 0x0 0x68 0x4d 0x8 0x2 0x81 0x91 ] Aug 2 17:58:31 asterisk1 kernel: ztx 13 bytes Aug 2 17:58:31 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0xf4 0x3a 0x8 0x2 0x80 0x68 0x5a 0xab 0x84 ] 11 bytes Aug 2 17:58:31 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf6 ] Aug 2 17:58:31 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:41 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x3b 0xfc 0x9 ] 6 bytes Aug 2 17:58:41 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf7 ] Aug 2 17:58:41 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:51 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x3b 0xfc 0x9 ] 6 bytes Aug 2 17:58:51 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf7 ] Aug 2 17:58:51 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:57 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0xf6 0x3a 0x8 0x2 0x26 0xb6 0x5 0x4 0x3 0x80 0x90 0xa3 0x18 0x3 0xa9 0x83 0x8c 0x6c 0x5 0x1 0x81 0x34 0x37 0x37 0x70 0x4 0x81 0x34 0x34 0x30 0x7d 0x2 0x91 0x81 0x9d 0x32 0x1 0x81 0xea 0xb2 ] 42 bytes Aug 2 17:58:57 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf8 ] Aug 2 17:58:57 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:57 asterisk1 kernel: cwain: card 1 TX [ 0x0 0x1 0x3a 0xf8 0x8 0x2 0xa6 0xb6 0x2 0x18 0x3 0xa9 0x83 0x8c ] Aug 2 17:58:57 asterisk1 kernel: ztx 14 bytes Aug 2 17:58:57 asterisk1 kernel: cwain: card 1 RX [ 0x0 0x1 0x1 0x3c 0x35 0x44 ] 6 bytes Aug 2 17:58:57 asterisk1 kernel: cwain: card 1 TX [ 0x0 0x1 0x3c 0xf8 0x8 0x2 0xa6 0xb6 0x1 0x1e 0x2 0x81 0x88 ] Aug 2 17:58:57 asterisk1 kernel: ztx 13 bytes Aug 2 17:58:57 asterisk1 kernel: cwain: card 1 RX [ 0x0 0x1 0x1 0x3e 0x27 0x67 ] 6 bytes --- snap --- Is ist possible to disable this? Regards Bastian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79XX and VLANS
Oh damn. I'm not using cisco switches, but a dell 3348 (I know, I know) No way to turn it off on the phone, then ? Julian. Erik Versaevel - Infopact Netwerkdiensten BV wrote: If you're using Cisco Switches: Logon to the switch and go to config mode int fa0/1 switchport access voice vlan untagged sometheing in that direction configures the CDP to set the phone to untagged frames. Julian Lyndon-Smith wrote: How ? Where ? I've been wanting to do this for ages, and never found an option to do so ! Please Please Please tell all. (I hate begging, but sometimes ) Julian. Eric Wieling aka ManxPower wrote: Matthew Boehm wrote: Hey gang, We have about 30 Cisco 79XX phones all running latest 7.5 SIP. We are also using all Cisco Switches and Routers. Everything works great except that when you reboot a phone it takes like 3-5 minutes for it to come up. The phones spend tons of time 'Configuring VLAN..' We don't run any VLANs. Is there some way to skip this? In the 'Network Settings' I have both 'Operational VLAN Id' and 'Admin VLAN Id' set to blank values. Disable CDP on the phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P V2 changes?
Hi, we also got one V2 TE405P card. It works fine now. At the moment we use for bridging the Pri to our old PBX. You must use zaptel, libri and asterisk v 1.0.8 or higher. We use 1.0.9 at the moment. zaptel: after make; make install i also executed make config. This copies the correct startup script to /etc/init.d/zaptel. Without this it also didn't worked for me. Master Abi wrote: Hi I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after putting in the upgraded board but did not change any conf, but the spans become active but will not come up. I guess I am missing something or are the any changes to the zaptel/libpri software that is required. I cannot find any info about this or does this new firmware only work with latest CVS. I am using 1.0.9 with 2.6.12 kernel Zapata Telephony Interface Registered on major 196 Found TE4XXP at base address fdfff000, remapped to f8928000 TE4XXP version c01a0164, burst ON, slip debug: OFF TE4XXP running with work queues. FALC version: 0005, Board ID: 00 Reg 0: 0x364e9400 Reg 1: 0x364e9000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a0164 Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE405P (2nd Gen) eth0: link up, 10Mbps, half-duplex, lpa 0x About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Unassigning channel 0/1! Unassigning channel 0/2! Unassigning channel 0/3! Unassigning channel 0/4! Unassigning channel 0/5! Unassigning channel 0/6! Unassigning channel 0/7! Unassigning channel 0/8! Unassigning channel 0/9! Unassigning channel 0/10! Unassigning channel 0/11! Unassigning channel 0/12! Unassigning channel 0/13! Unassigning channel 0/14! Unassigning channel 0/15! Unassigning channel 0/16! Unassigning channel 0/17! Unassigning channel 0/18! etc... This was working for 10 months before the upgrade. Master ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Billion BRI PCI card
I'm looking at experimenting with asterisk with an ISDN BRI and ISDN phones (since I have these already). I saw that the Billion card was cheap and could be used in either TE or NT modes. I have the following question which I couldn't answer by reading through the manual. Maybe someone has experience of using this card and can help me out. when using in NT mode does the card require additional power or is it able to supply enough power by itself to the S0 bus? thanks for any help, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billion BRI PCI card
Hello, when using in NT mode does the card require additional power or is it able to supply enough power by itself to the S0 bus? You will need an additional power (for example http://shop.beronet.com/product_info.php/products_id/48). Best regards Blaise ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billion BRI PCI card
On Friday 12 August 2005 09:43, John Fawcett wrote: when using in NT mode does the card require additional power or is it able to supply enough power by itself to the S0 bus? I don't know the exact specifics about the Billion card, but I have a setup where I have an extra NTBA connected to the ISDN card (S0 bus coming from ISDN card into NTBA, cable needs to be crossed) and then my ISDN phones are connected to the S0 bus coming out of the NTBA. I assume that the Billion card works in a similar fashion and doesn't have the necessary power to directly connect phones to it. I hope that this helps you a little. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Status of app_sms in 1.0.9
Hello all, can anybody how usable app_sms is? I want to use it in england (but not with the BT) and in Germany. Is this possible with * 1.0.9 and either with PRI lines or with simple ISDN and an AVM Fritz! Card?? Thx in advance, Tobias Wolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 79XX and VLANS
I have a * system at home with mainly 7960's. I run a trunk into each of the phones, it is the only way to get cos bits set correctly from the phone into the switch. I then translate the cos bits into DSCP for layer 3. I have separate vlans for voice and data, * server has 2 nic's one in each vlan. It has the side effect that 'Configuring VLAN' takes about 5 seconds. Peter -Original Message- From: Matthew Boehm [mailto:[EMAIL PROTECTED] Sent: 11 August 2005 20:26 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco 79XX and VLANS Hey gang, We have about 30 Cisco 79XX phones all running latest 7.5 SIP. We are also using all Cisco Switches and Routers. Everything works great except that when you reboot a phone it takes like 3-5 minutes for it to come up. The phones spend tons of time 'Configuring VLAN..' We don't run any VLANs. Is there some way to skip this? In the 'Network Settings' I have both 'Operational VLAN Id' and 'Admin VLAN Id' set to blank values. Any ideas? Thanks, Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ __ This email has been scanned for all viruses by the Star Internet Virus Screen. The service is provided in partnership with MessageLabs, the email security company. For more information on a higher level of virus protection visit www.star.net.uk __ __ This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Install just to play with experiment
Is there a significant difference between the features set of [EMAIL PROTECTED] and regular asterisk? (important missing features?) About the cautions.. You're not kidding.. It blew out the hard drive without so much as a you sure you want to do this?. Unfortunately I figured this out too late.. fortunately it was only a test system that won't take long to return to it's previous state.. I may even use it as Asterisk's permanent home.. A dual P3 500 server class intel board.. - Michael From: Tom Rymes [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thu, 11 Aug 2005 18:45:55 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Install just to play with experiment Considering your description, I think that you should definitely try out [EMAIL PROTECTED] It includes AMP already configured and working, and makes it very easy to configure and set up a SOHO system. you can even set it up to make and receive phone calls by configuring an account at Teliax, VoipJet (outgoing only, no incoming as of yet...), or Broadvoice. Go to asteriskathome.sourceforge.net and dowload the ISO fiile. As mentioned earlier, BE CAREFUL!!! This CD will format your hard drive, wipe out all data, and install a new CentOS system without asking for your say-so. Tom On Aug 11, 2005, at 6:37 PM, Francesco Peeters wrote: On Fri, August 12, 2005 0:25, Doug Lytle said: Michael Jones wrote: Is there a quick configuration that can be put into place to simply experiment with the system (like create a couple of extensions wit Xlite and make a internal phone system in my office that doesn't really go out anywhere - or maybe connects to a voip provider later)? If you compile from source you can do a make samples . Doug And otherwise, you can try Asterisk @ Home, but be warned that it installs a complete new system, so do not try it on an existing machine you do not wish to be wiped clean first! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Supervised transfer problem with BudgetTone
The VoIP Connection a écrit : Nicolas, Just did some quick testing and the instructions are incorrect. You need to press transfer to complete the transfer instead of the second flash. This actually makes more sense. Attended and regular transfer both work perfectly with the following settings: Enable Call Features: Yes Disable call Waiting: No Send Flash event: No DTMF should be whatever * is set to, but in-band won't work properly if your codec is anything other than U-Law. By the way, the newest firmware also makes the long overdue conference feature work properly. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Nicolas Schmerber [mailto:[EMAIL PROTECTED] Sent: Thursday, August 11, 2005 10:41 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Supervised transfer problem with BudgetTone The VoIP Connection a écrit : Section 4.3.7.2 from the Bugetone Manual: The user can transfer an active call to a third party with announcement. The user presses the “flash” button and hears a dial tone, then dial the 3rd party’s phone number followed by pressing send button. If the call is answered, press “flash” to complete the transfer operation, if the call is not answered, pressing “flash” button to resume the original call. Notes: • If attended Transfer fails, the BudgeTone phone will ring the user to remind that another party is still on the call, the user can then pick up the call using handset or speaker. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Nicolas Schmerber [mailto:[EMAIL PROTECTED] Sent: Thursday, August 11, 2005 5:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Supervised transfer problem with BudgetTone [EMAIL PROTECTED] a écrit : On Thu, 11 Aug 2005, Nicolas Schmerber wrote: All the features I need work just not one : the supervised call transfers. I know there are a lot of posts about that, but none gave me the correct answer (unless I missed it). Hi, You'll need to switch to the CVS-HEAD version of Asterisk in order to have supervised transfers. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users When looking at a recent firmware changelog of Grandstream , it says BT should support supervised transfer, so shouldnt it work ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tried this manipulation a few minutes ago : A calls B , B pushes flash button ( A is waiting with a mp3 played) B calls C pressing Send ; C answers B presses flash button again ; C is so on hold (with a mp3 played) B hangs up But A and C arent in connect ; the phoneof B rings ( to tell someone is in wait : A) So it seems to fail What should i put in grandstream config for the next item : /Enable Call Features: Y/ N ? //Disable Call-Waiting: Y/N ? //Send DTMF: / in-audio / via RTP (RFC2833) / via SIP INFO /Send Flash Event: Y / N ? / Any others Ideas ?. Thx Nicolas S. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you all, now it works The last method (grandsteam manual but with transfer key instead) was the right Thanks Nicolas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with calling Perl AGI interface
I'll second that. Make sure your script is in /var/lib/asterisk/agi-bin and you have the right permissions on it. I really just wanted to reply to your post though to congraduate you, Dan Marino, on your recent induction into the Pro Football Hall of Fame ;) Sorry, wrong Dan Marino! -Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with calling Perl AGI interface
On 8/10/05, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Dan Marino wrote: I have installed the Perl library from http://asterisk.gnuinter.net/asterisk-perl and am wondering how I reference agi-test.agi from extensions.conf I have added exten = s,1,AGI,agi-test.agi but that doesn't seem to do it. Is there a certain directory .agi files should be, is that the problem? Depending on your asterisk install, the agi-bin directory can be somewhere like /var/lib/asterisk/agi-bin or /usr/share/asterisk/agi-bin locate agi-bin is your friend :) Cheers, Jean-Michel. Thanks! I found the agi-bin it is working -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP_HEADER
Anybody knows, how to use the SIP_HEADER function? Thanks Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7920 boot causes 7940 to release DHCP lease
On Thu, Aug 11, 2005 at 07:12:32PM +0100, Mark Thorpe wrote: I have been trying to solve a problem wherby when I boot a cisco 7920 my 7940 seeks a new IP and the dhcpd log shows it released its existing IP. In searching for the solution I notice there were 2 messages on this list in Aug Sep 2004 which raised the problem, but I could not find any answer was posted. I've noticed the same issue. Haven't tried it with 79[46]0's 7.5 firmware. A workaround would be to give a static addres to your 7920. Let me know if you find a solution. -- I have no special talents. I am only passionately curious. --Albert Einstein ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Question: Building an Asterisk system toreplace an old PBX but using existing phone
Michael Boger Jr wrote: Sean, What kind of hotel do you have? Some PMS vendors require the call accounting and check-in interfaces to their system. I am not aware that asterisk supports these serial interfaces. No they have no call accounting etc as such everything is done manually. I will work out printing at a later stage Sean -- ICQ: 679813FidoNet: 2:263/950 Jabber: [EMAIL PROTECTED] AOL: tcobone Vodafone Messenger: thecivvie smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Sendmail question
Hi, all I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. Using Ethereal, I can see that messages are leaving my network, but receipeint mail server never replies back. As a result, mail delivery is timed out. I got a book on sendmail and it looks quite complex. It will take quite a bit of time to find out what is going on. I am using FC3 and sendmail uses default configuration. Is teher a quick tweak I can do to get it to work? May be someone can suggest another mail program that is easier to setup? Messages sent from command line behave same way as ones sent from asterisks, so it is definetely a sendmail configuration issue. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Sendmail question
Default sendmail should work. Try to test sendmail from console. Some SMTP maybe block the email. run mail to see if your email is bounced back. Kun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rudolf Ladyzhenskii Sent: Friday, August 12, 2005 5:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OT: Sendmail question Hi, all I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. Using Ethereal, I can see that messages are leaving my network, but receipeint mail server never replies back. As a result, mail delivery is timed out. I got a book on sendmail and it looks quite complex. It will take quite a bit of time to find out what is going on. I am using FC3 and sendmail uses default configuration. Is teher a quick tweak I can do to get it to work? May be someone can suggest another mail program that is easier to setup? Messages sent from command line behave same way as ones sent from asterisks, so it is definetely a sendmail configuration issue. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Sendmail question
Thanks for reply. I would expect it to work too, but it does not. I tried to send mail from console -- same result. Messages are just sitting in teh queue. sendmail times out sending them. Mail does not bounce. Rudolf - Original Message - From: Wei Kun [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, August 12, 2005 7:36 PM Subject: RE: [Asterisk-Users] OT: Sendmail question Default sendmail should work. Try to test sendmail from console. Some SMTP maybe block the email. run mail to see if your email is bounced back. Kun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rudolf Ladyzhenskii Sent: Friday, August 12, 2005 5:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OT: Sendmail question Hi, all I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. Using Ethereal, I can see that messages are leaving my network, but receipeint mail server never replies back. As a result, mail delivery is timed out. I got a book on sendmail and it looks quite complex. It will take quite a bit of time to find out what is going on. I am using FC3 and sendmail uses default configuration. Is teher a quick tweak I can do to get it to work? May be someone can suggest another mail program that is easier to setup? Messages sent from command line behave same way as ones sent from asterisks, so it is definetely a sendmail configuration issue. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billion BRI PCI card
when using in NT mode does the card require additional power or is it able to supply enough power by itself to the S0 bus? You will need an additional power (for example http://shop.beronet.com/product_info.php/products_id/48). This is for the 4xBRI or 8xBRI cards from Beronet. The Billion 1xBRI has net the necessary connectors on it. You will need network terminator (NTBA) a selfmase power supply if the phones on the BRI have no own power supply. Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Sendmail question
I think, I just sorted i out. I have to run sendmail with optiosn -bm to be a mail sender. Without it, it seems that sendmail is trying to use outside server for delivery. Without valid username, this will not work... Rudolf - Original Message - From: Rudolf Ladyzhenskii [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 12, 2005 7:32 PM Subject: Re: [Asterisk-Users] OT: Sendmail question Thanks for reply. I would expect it to work too, but it does not. I tried to send mail from console -- same result. Messages are just sitting in teh queue. sendmail times out sending them. Mail does not bounce. Rudolf - Original Message - From: Wei Kun [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, August 12, 2005 7:36 PM Subject: RE: [Asterisk-Users] OT: Sendmail question Default sendmail should work. Try to test sendmail from console. Some SMTP maybe block the email. run mail to see if your email is bounced back. Kun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rudolf Ladyzhenskii Sent: Friday, August 12, 2005 5:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OT: Sendmail question Hi, all I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. Using Ethereal, I can see that messages are leaving my network, but receipeint mail server never replies back. As a result, mail delivery is timed out. I got a book on sendmail and it looks quite complex. It will take quite a bit of time to find out what is going on. I am using FC3 and sendmail uses default configuration. Is teher a quick tweak I can do to get it to work? May be someone can suggest another mail program that is easier to setup? Messages sent from command line behave same way as ones sent from asterisks, so it is definetely a sendmail configuration issue. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Sendmail question
how come you said mail is send out but still in the queue? Does it send out or not? Kun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rudolf Ladyzhenskii Sent: Friday, August 12, 2005 5:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: Sendmail question Thanks for reply. I would expect it to work too, but it does not. I tried to send mail from console -- same result. Messages are just sitting in teh queue. sendmail times out sending them. Mail does not bounce. Rudolf - Original Message - From: Wei Kun [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, August 12, 2005 7:36 PM Subject: RE: [Asterisk-Users] OT: Sendmail question Default sendmail should work. Try to test sendmail from console. Some SMTP maybe block the email. run mail to see if your email is bounced back. Kun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rudolf Ladyzhenskii Sent: Friday, August 12, 2005 5:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OT: Sendmail question Hi, all I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. Using Ethereal, I can see that messages are leaving my network, but receipeint mail server never replies back. As a result, mail delivery is timed out. I got a book on sendmail and it looks quite complex. It will take quite a bit of time to find out what is going on. I am using FC3 and sendmail uses default configuration. Is teher a quick tweak I can do to get it to work? May be someone can suggest another mail program that is easier to setup? Messages sent from command line behave same way as ones sent from asterisks, so it is definetely a sendmail configuration issue. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billion BRI PCI card
I'm looking at experimenting with asterisk with an ISDN BRI and ISDN phones (since I have these already). I saw that the Billion card was cheap and could be used in either TE or NT modes. I have the following question which I couldn't answer by reading through the manual. Maybe someone has experience of using this card and can help me out. when using in NT mode does the card require additional power or is it able to supply enough power by itself to the S0 bus? The cards feeds no power to the s0. If the phone has its own power supply normally it will work without external power supply. Otherwise you will need a network terminator (NTBA) or a selfmade power injector. Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Sendmail question
Old messages are in the queue. I can see sendmail is trying to talk to the remote mail server, but never gets a responce and times out. So message stays in the queue. Rudolf - Original Message - From: Wei Kun [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, August 12, 2005 7:50 PM Subject: RE: [Asterisk-Users] OT: Sendmail question how come you said mail is send out but still in the queue? Does it send out or not? Kun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rudolf Ladyzhenskii Sent: Friday, August 12, 2005 5:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: Sendmail question Thanks for reply. I would expect it to work too, but it does not. I tried to send mail from console -- same result. Messages are just sitting in teh queue. sendmail times out sending them. Mail does not bounce. Rudolf - Original Message - From: Wei Kun [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, August 12, 2005 7:36 PM Subject: RE: [Asterisk-Users] OT: Sendmail question Default sendmail should work. Try to test sendmail from console. Some SMTP maybe block the email. run mail to see if your email is bounced back. Kun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rudolf Ladyzhenskii Sent: Friday, August 12, 2005 5:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OT: Sendmail question Hi, all I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. Using Ethereal, I can see that messages are leaving my network, but receipeint mail server never replies back. As a result, mail delivery is timed out. I got a book on sendmail and it looks quite complex. It will take quite a bit of time to find out what is going on. I am using FC3 and sendmail uses default configuration. Is teher a quick tweak I can do to get it to work? May be someone can suggest another mail program that is easier to setup? Messages sent from command line behave same way as ones sent from asterisks, so it is definetely a sendmail configuration issue. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Sendmail question
On Friday 12 August 2005 11:46, Rudolf Ladyzhenskii wrote: Old messages are in the queue. I can see sendmail is trying to talk to the remote mail server, but never gets a responce and times out. So message stays in the queue. You should try and deliver them yourself. It sounds to me like sendmail is configured to relay messages using a smart host (the one it's trying to talk to). Do you maybe need to log in to that host in order to send mail? You can try this out yourself: set the DNS name of the mail server it relays it to in /etc/hosts to your local host, don't start sendmail as a server daemon (shouldn't bind to port 25) and do a nc -lt -p 25 and look what it's trying to do when sending mail via the command line. Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] v92 modems
Hello, Is it possible to use v92 ( a few chipsets version ) modem as FXO PCI modules ? While googling I found some postings on the subject. Thanks Varun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call recording, monitor soxmix in Asterisk 1.0.9
Hi, Monitor and soxmix (m option) work fine in CVS Head, not in Asterisk 1.0.9, as the Wiki says.http://www.voip-info.org/tiki-index.php?page=Monitor+setup+sample Anyway I am wondering why asterisk 1.0.9console shows on Hang up: "monitor executing ( nice -n 19 soxmix "//var/spool/asterisk/monitor/45/47-20050812-113631-in.wav" "//var/spool/asterisk/monitor/45/47-20050812-113631-out.wav" "//var/spool/asterisk/monitor/45/47-20050812-113631.wav" rm -f "//var/spool/asterisk/monitor/45/47-20050812-113631-"* ) "and It doesn't work, I mean,what I find in /monitor/45 are the two -in and -out files.But it is curiousthatif you type the same command (nice -n 19 etc..) from the command line, It does work. Doesanybody know why? Kind regards, Victor. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE405P / TE410P with 2nd generation firmware field upgradable?
hi, after stumbling over the compile time flag in zaptel and after reading the new features of the 2nd generation firmware of the TE405P/TE410P, i was wondering if the cards are capable of upgrading the firmware in field? regards frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] v92 modems
On Friday 12 August 2005 06:42, [EMAIL PROTECTED] wrote: Hello, Is it possible to use v92 ( a few chipsets version ) modem as FXO PCI modules ? Short answer: no. Longer answer: perhaps, but you're on your own. Your googling efforts should have shown you that. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] v92 modems
[EMAIL PROTECTED] wrote: Hello, Is it possible to use v92 ( a few chipsets version ) modem as FXO PCI modules ? While googling I found some postings on the subject. Thanks Varun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, Yes, Modem with an Ambient MD3200, Model # : AMI-IA92/IE92 The Digium products has great performance. Buy Digium products and support Asterisk! Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P V2 changes?
Are you using Redhat/Fedora? If I remember those init scripts is for Redhat/Fedora. I am using gentoo. Did you make any modifications to wct4xxp.c. or pass any parameters to zaptel. I see there is a #define SUPPORT_GEN1 in to wct4xxp.c which I commented out, but it made no difference. ztcfg seems to where the channels become unassigned. Thanks again. Kib Eki wrote: Hi, we also got one V2 TE405P card. It works fine now. At the moment we use for bridging the Pri to our old PBX. You must use zaptel, libri and asterisk v 1.0.8 or higher. We use 1.0.9 at the moment. zaptel: after make; make install i also executed make config. This copies the correct startup script to /etc/init.d/zaptel. Without this it also didn't worked for me. Master Abi wrote: Hi I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after putting in the upgraded board but did not change any conf, but the spans become active but will not come up. I guess I am missing something or are the any changes to the zaptel/libpri software that is required. I cannot find any info about this or does this new firmware only work with latest CVS. I am using 1.0.9 with 2.6.12 kernel Zapata Telephony Interface Registered on major 196 Found TE4XXP at base address fdfff000, remapped to f8928000 TE4XXP version c01a0164, burst ON, slip debug: OFF TE4XXP running with work queues. FALC version: 0005, Board ID: 00 Reg 0: 0x364e9400 Reg 1: 0x364e9000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a0164 Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE405P (2nd Gen) eth0: link up, 10Mbps, half-duplex, lpa 0x About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Unassigning channel 0/1! Unassigning channel 0/2! Unassigning channel 0/3! Unassigning channel 0/4! Unassigning channel 0/5! Unassigning channel 0/6! Unassigning channel 0/7! Unassigning channel 0/8! Unassigning channel 0/9! Unassigning channel 0/10! Unassigning channel 0/11! Unassigning channel 0/12! Unassigning channel 0/13! Unassigning channel 0/14! Unassigning channel 0/15! Unassigning channel 0/16! Unassigning channel 0/17! Unassigning channel 0/18! etc... This was working for 10 months before the upgrade. Master ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billion BRI PCI card
Christian Victor wrote: The cards feeds no power to the s0. If the phone has its own power supply normally it will work without external power supply. Otherwise you will need a network terminator (NTBA) or a selfmade power injector. Christian Christoph Eicke wrote: I don't know the exact specifics about the Billion card, but I have a setup where I have an extra NTBA connected to the ISDN card (S0 bus coming from ISDN card into NTBA, cable needs to be crossed) and then my ISDN phones are connected to the S0 bus coming out of the NTBA. I assume that the Billion card works in a similar fashion and doesn't have the necessary power to directly connect phones to it. I hope that this helps you a little. thanks to everyone for all the replies. I have a doubt about how to connect the NTBA, since it has a U interface and an S0 interface with two sockets. Would I connect the Billion card to one of the S0 sockets on the NTBA (via a crossover cable) and then the telephone to the other S0 socket. I assume that I don't connect anything to the U interface of the NTBA. Is that correct? Thanks, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P / TE410P with 2nd generation firmware field upgradable?
On Friday 12 August 2005 06:59, Frank Sautter wrote: after stumbling over the compile time flag in zaptel and after reading the new features of the 2nd generation firmware of the TE405P/TE410P, i was wondering if the cards are capable of upgrading the firmware in field? Unfortunately not. It's a configuration PROM, not EEPROM or FLASH. I am pretty sure that Digium has an upgrade program in place though. It's best to contact them directly for these types of inquiries instead of the list. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P / TE410P with 2nd generation firmware field upgradable?
Frank Sautter schrieb: hi, after stumbling over the compile time flag in zaptel and after reading the new features of the 2nd generation firmware of the TE405P/TE410P, i was wondering if the cards are capable of upgrading the firmware in field? It is said so - but I don't believe it. ;-) Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billion BRI PCI card
John Fawcett wrote: I have a doubt about how to connect the NTBA, since it has a U interface and an S0 interface with two sockets. Would I connect the Billion card to one of the S0 sockets on the NTBA (via a crossover cable) and then the telephone to the other S0 socket. I assume that I don't connect anything to the U interface of the NTBA. Is that correct? That's correct. You can either use a crossover cable and plug it into one of the S0 sockets or cut an ISDN cable in half and connect it to the S0 plugs inside the NTBA. Andreas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P / TE410P with 2nd generation firmware field upgradable?
Short answer: NO Long answer: you have to send it to Digium for them to do an upgrade, they don't have an official process for this yet and won't give you a price, I have called and asked them many times. They also mention upgrades from your 405/410 to a 406/411 are available too, but again no specifics. Supposedly if you have a card with the 2nd gen firmware on it you can upgrade to the third gen firmware, whenever it would come out, in the field. MATT--- On 8/12/05, Christian Victor [EMAIL PROTECTED] wrote: Frank Sautter schrieb: hi, after stumbling over the compile time flag in zaptel and after reading the new features of the 2nd generation firmware of the TE405P/TE410P, i was wondering if the cards are capable of upgrading the firmware in field? It is said so - but I don't believe it. ;-) Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Voice mail to nortel PBX Option 11
Easily doable. I've done it twice now. Problem is that your users will never know they have messages waiting. Install a T1/E1 card into the * box and then use a T1 cross-over cable between the 2 boxes. Create a dialplan on the Meridian that points calls to the VM out over the new E1. As for forwarding the calls when busy or no answer, that's a little more tricky. You'll have to come up with some rules and numbers to allow the Meridian to decide what to do with those calls. In my case I wrote a forward on no answer and a forward on busy rule for every phone that needed VM. When you called ext 200 the call was sent to mailbox 2200 on the *. Users will have to get into the habit of calling the VM to check if there's messages. Hope that helps. Mark craz sead wrote: Hi all, Could somebody help me, i wanna connect asterisk for voice mail in the existing nortel pbx option 11 using e1 card ? anyone have a clue ? please help the conf. file thank all __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Install just to play with experiment
AAH uses the latest released version of *. This is somewhat different to the CVS but can be considered stable for the purposes of production. It also has the added benefits of a web front end to configure it as well as some other nice webby features to make life easier when running it. The CVS and the Stable versions have to be hand crafted when dealing with the config files etc. I have run both in my time. Currently I'm using AAH which for the most part is fine as long as I don't want to do anything that the web front end won't let me do. Once I start editing the config files the system falls apart. Horses for courses I think. Mark Michael Jones wrote: Is there a significant difference between the features set of [EMAIL PROTECTED] and regular asterisk? (important missing features?) About the cautions.. You're not kidding.. It blew out the hard drive without so much as a you sure you want to do this?. Unfortunately I figured this out too late.. fortunately it was only a test system that won't take long to return to it's previous state.. I may even use it as Asterisk's permanent home.. A dual P3 500 server class intel board.. - Michael From: Tom Rymes [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thu, 11 Aug 2005 18:45:55 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Install just to play with experiment Considering your description, I think that you should definitely try out [EMAIL PROTECTED] It includes AMP already configured and working, and makes it very easy to configure and set up a SOHO system. you can even set it up to make and receive phone calls by configuring an account at Teliax, VoipJet (outgoing only, no incoming as of yet...), or Broadvoice. Go to asteriskathome.sourceforge.net and dowload the ISO fiile. As mentioned earlier, BE CAREFUL!!! This CD will format your hard drive, wipe out all data, and install a new CentOS system without asking for your say-so. Tom On Aug 11, 2005, at 6:37 PM, Francesco Peeters wrote: On Fri, August 12, 2005 0:25, Doug Lytle said: Michael Jones wrote: Is there a quick configuration that can be put into place to simply experiment with the system (like create a couple of extensions wit Xlite and make a internal phone system in my office that doesn't really go out anywhere - or maybe connects to a voip provider later)? If you compile from source you can do a make samples . Doug And otherwise, you can try Asterisk @ Home, but be warned that it installs a complete new system, so do not try it on an existing machine you do not wish to be wiped clean first! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] v92 modems
Yes, but your results may vary. Apparently some people have problems with clone cards (aka regular modems), dropping calls, and having echos. (Then again some people have reported no problems at all). E-bay is a good source for these. You can also check out this list with more information about Asterisk clone cards here: http://www.voip-info.org/tiki-index.php?page=X100P+clone On 8/12/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Friday 12 August 2005 06:42, [EMAIL PROTECTED] wrote: Hello, Is it possible to use v92 ( a few chipsets version ) modem as FXO PCI modules ? Short answer: no. Longer answer: perhaps, but you're on your own. Your googling efforts should have shown you that. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Voice mail to nortel PBX Option 11
Isn't it possible to turn on MWI via background terminal ? In that case an application needs to do this via serial interface. best regards Hans Users will have to get into the habit of calling the VM to check if there's messages. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
Kevin P. Fleming wrote: Geoff Manning wrote: The TE110P card in the Asterisk server is set as the sync source: span=1,1,0,d4,ami em=1-24 That is incorrect. You have your span configured to recover timing from the T1 and use that as the source for the card. If you want this span to be clocked using the onboard clock on the board, you must use: span=1,0,0,d4,ami I though setting it at 0 was to tell asterisk not to be the source of the timing. When I set it at 1 I get slip errors inplace of the blue alarms. I must have had my logic backwards. Thanks, Geoff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] re: call load balancing
Anton Krall [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] intruder]# ps afx|more PID TTY STAT TIME COMMAND 1 ?S 0:08 init 2 ?SW 0:00 [keventd] 3 ?SW 0:00 [kapmd] 4 ?SWN0:00 [ksoftirqd_CPU0] 9 ?SW 0:00 [bdflush] No priorities.. Am I missing something? Try ps alx (Look at the NI column). Also see man ps. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
Peter Svensson wrote: A blue alarm sounds really strange. That indicates that the remote end (asterisk) in this case does not want to play at all. On a T1 it is sent as a continous series of unframed 1:s. I am not sure if asterisk ever sends a blue alarm (Alarm Indication Signal). Receiving a blue alarm is indicative of a serious problem. There should not be any audio at that time, since the blue alarm is actually a long unframed signal. Well that would explain the choppy/stuttering sound we get on these calls since there is no audio during those error transmissions. According to Kevin's reply I had my timing logic backwards. Should I be using any other timing settings on the Asterisk side?? The tech for our legacy PBX says that the PBX will not provide any timing. Thanks, Geoff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Sendmail question
I think it would help if you sent an excerpt from your maillog. Cheers Wayne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] yahoo voice
Only because I dont want to install yet another IM to my existing Skype and MSN has anyone tried the new yahoo voice? http://www.smh.com.au/news/technology/scramble-to-find-voice-on-the-web/2005/08/12/1123353481827.html http://messenger.yahoo.com/feat_voice.php Any thoughts? Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] USB handset wanted
Give me a handset with a ringer and hook that both work under Linux and I will buy 30 pieces on the spot! No kidding.. Bill McCready (PCPhoneline.com) wrote: We are planning to develop versions of our USB based phone and gateway products for Linux. The plan is to make them will work like regular phones exactly like our Windows versions do including physically ringing loudly on incoming calls. Which versions of Linux are the most popular at the moment in the workplace so we can decide which one to focus our energies on first? Best regards, Bill McCready PCPhoneline.com - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 11, 2005 1:38 PM Subject: Re: [Asterisk-Users] USB handset wanted Ondrej Valousek wrote: Matt, You have forgotten the ringer. In fact, I don't care that much about LCD buttons. I want to use it with something like X-lite. Initially, I used machine builtin soundcard with X-Lite (worked well) but then I realized that if the phone is supposed to compete with the standard analog phone, it must have a working ringer. Fair enough. From what I see I suppose that every handset with builtin ringer must be recongized to the OS as 2 USB soundcards - one for speaker/mike, the second as a ringer. The ones I have worked with have a seperate ringer that just takes an int to decide which ringtone to play. I.E. it is not shown as a soundcard. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adtran TSU 600
I got a good deal on one of these channel banks loaded with 24 FXS ports. I know 24 seems pretty overkill for a home user, but I got this shipped cheaper than I could have gotten a TDM400P w/ 1 FXS port. I've read that these are compatible w/ asterisk, but can they be used w/o a T1?? (I'm not really sure how * is connected to the channel bank). Would I have to have a T100P (whatever the new model is.. T1/E1 selectable.. blah blah) and a T1 xover cable? (If so, suddenly the deal just got more expensive) I have it running already for a about a year with no problems what so ever. U need T1 (t100p is good) card for asterisk and crossover T1 cable. Bartosz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Possibly bad FXS module in TDM400P?
I've got the latest zaptel and cvs asterisk software loaded on my phone server running FC3. And yes, It's fully updated and udev is setup correctly. I've got a TDM400P with one FXS and one FXO module installed. When I load zaptel and wctdm and run ztcfg -vvv, I get this: [EMAIL PROTECTED] ~]# ztcfg -vvv ZT_CHANCONFIG failed on channel 1: No such device or address (6) Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. Regardless of my configuration, it should show up in ztcfg, right? Here's my: ** zaptel.conf loadzone = us defaultzone = us fxoks=1 fxsks=2 ** zapata.conf [trunkgroups] [channels] language=en context=default switchtype=national signalling=fxo_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=yes canpark=yes cancallforward=yes callreturn=yes mailbox=2500 echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no context=default channel = 1 context=incoming signalling=fxs_ks group=2 channel = 2 Thanks for your help. Jeff Borders ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP301 and 501 with asterisk...
Chris, Maybe you could write a generic config file and post it to the wiki? MAC address = Server IP = xxx.xxx.xxx.xxx Username = user password = pass, Extension = 100 Just a thought Tom On Aug 11, 2005, at 10:15 PM, Chris Mason wrote: Shaun Bolling wrote: Jonathan, did you have any problem getting your polycom 301 to work with asterisk. I purchase two of them for testing. I have been trying for two days now to get them to call one another, with no luck. My software phones work fine. In my asterisk log I get a error message like Failed to authenticate user. Did you have this problem? Send me the info and I will write you a config file for one phone: MAC of phone serverip username (for ftp account) password (for ftp account) Extension number codec preference I'm presuming you are using DHCP to allocate TFTP setting, SNTP server, phone IP, Gateway. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: call does not hangup after client quits
Hi Jeremy, thanks for this, it looks like just what i was after. i've not finished testing it but it seems to do what i need. Steve On Tue, 9 Aug 2005, Jeremy Gault wrote: Steve, If I am understanding your situation correctly (i.e. you are using a SIP client and then forcibly disconnecting/shutting it off during a call) you may want to look at your sip.conf for a setting called rtptimeout. This may do exactly what you want. When on a SIP call, and you disconnect/shut off your client (without properly hanging up first) then (obviously) * does not receive a SIP message saying the call has ended. However, the RTP (audio) stream will stop. The rtptimeout setting lets you define a time period that after x seconds of no audio packets, it's assumed the SIP client has gone away and the call should be terminated. Jeremy Stephen J. Wilcox wrote: Hello, can anyone help with my problem below, searching doesnt show any results.. thanks Steve On Wed, 3 Aug 2005, Stephen J. Wilcox wrote: Hi, I'm seeing a problem where if I place a call, then forcibly quit or turn off the client the call stays active. The frames counters stop so its apparent the client has gone away but the call remains active. Asterisk is CVS-HEAD 23-Jun-05 What is supposed to happen in this scenario? thanks Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
Kevin P. Fleming wrote: Geoff Manning wrote: The TE110P card in the Asterisk server is set as the sync source: span=1,1,0,d4,ami em=1-24 That is incorrect. You have your span configured to recover timing from the T1 and use that as the source for the card. If you want this span to be clocked using the onboard clock on the board, you must use: span=1,0,0,d4,ami OK. So I changed it to: span=1,0,0,d4,ami And the Blue Alarms are still occurring but now in conjunction with Slip errors. I feel like I am on the right track though. Another thing that is happening now is every once and a while all the channels on the span will open up and then timeout like so: -- Starting simple switch on 'Zap/1-1' -- Starting simple switch on 'Zap/2-1' -- Starting simple switch on 'Zap/3-1' -- Starting simple switch on 'Zap/4-1' -- Starting simple switch on 'Zap/5-1' -- Starting simple switch on 'Zap/6-1' -- Starting simple switch on 'Zap/7-1' -- Starting simple switch on 'Zap/8-1' -- Starting simple switch on 'Zap/9-1' -- Starting simple switch on 'Zap/10-1' -- Starting simple switch on 'Zap/11-1' -- Starting simple switch on 'Zap/12-1' -- Starting simple switch on 'Zap/13-1' -- Starting simple switch on 'Zap/14-1' -- Starting simple switch on 'Zap/15-1' -- Starting simple switch on 'Zap/16-1' -- Starting simple switch on 'Zap/17-1' -- Starting simple switch on 'Zap/18-1' -- Starting simple switch on 'Zap/19-1' -- Starting simple switch on 'Zap/20-1' -- Starting simple switch on 'Zap/21-1' -- Starting simple switch on 'Zap/22-1' -- Starting simple switch on 'Zap/23-1' Then they will timeout and hangup. I was on a call when this happened (Zap/24-1) and it didn't hang me up. But sometimes it will terminate my call when it happens. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_voicemail.c still looking for config fileeven I try to configure the voicemail from database.
Wei Kun wrote: mysql select * from extensions_table; ++--+---+--+---++ | id | context | exten | priority | app | appdata| ++--+---+--+---++ | 1 | from-sip | 2000 |1 | Dial | SIP/2000|20| | 2 | from-sip | 2000 |2 | Voicemail | u2000 | You need to call voicemail app with a context. Change the line above to show [EMAIL PROTECTED] -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] list in asterisk cli is getting too long
Hilton Williams wrote: - Original Message - From: Ronald Wiplinger To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, August 12, 2005 6:04 AM Subject: [Asterisk-Users] list in asterisk cli is getting too long How can I use something like|morein CLI ? The lists are getting too long, like sip show users You can also use sip show peers like pattern to truncate the list. Ex: sip show peers like 300 will only show peers whos username starts with or contains 300. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Install just to play with experiment
[EMAIL PROTECTED] is not a feature limited version of asterisk designed for home users. It is actually a collection of various software projects, including Asterisk stable, the AMP Web interface, SugarCRM, a Cisco phone configuration editor, plus ConfigEdit and a bunch of other stuff, all nicely tied together. On the other hand, Asterisk by default is a blank slate and there are extremely few limitations on how you can configure it to meet your needs. Extreme flexibility is the rule, and many do not like that GUIs (AMP included) necessarily place limits on this flexibility. As a result, it is harder to make more complex customizations. However, I think that the average user will actually find that AMP brings some order to the chaos that is Asterisk. Rather than limiting their freedom, many users see it as saving them from reinventing the wheel and having to learn the minutiae of Asterisk configuration. In addition, if you are willing to invest some time studying the inner workings of AMP, you can figure out how it works and place custom programming in the /etc/asterisk/ extensions_custom.conf file. In the end, it depends on what type of user you are, and what you want to accomplish. If you want to develop an easy to use and configure replacement for a traditional PBX that does not require you to really get into the nuts and bolts of dilaplans, etc, then I think [EMAIL PROTECTED] is for you. If you want to get into more complex things such as connecting databases to your system to let customers access their data interactively, etc, then you probably want to avoid the GUI, because it will limit your flexibility. As with most everything else, it depends! Tom On Aug 12, 2005, at 4:37 AM, Michael Jones wrote: Is there a significant difference between the features set of [EMAIL PROTECTED] and regular asterisk? (important missing features?) About the cautions.. You're not kidding.. It blew out the hard drive without so much as a you sure you want to do this?. Unfortunately I figured this out too late.. fortunately it was only a test system that won't take long to return to it's previous state.. I may even use it as Asterisk's permanent home.. A dual P3 500 server class intel board.. - Michael From: Tom Rymes [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thu, 11 Aug 2005 18:45:55 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Install just to play with experiment Considering your description, I think that you should definitely try out [EMAIL PROTECTED] It includes AMP already configured and working, and makes it very easy to configure and set up a SOHO system. you can even set it up to make and receive phone calls by configuring an account at Teliax, VoipJet (outgoing only, no incoming as of yet...), or Broadvoice. Go to asteriskathome.sourceforge.net and dowload the ISO fiile. As mentioned earlier, BE CAREFUL!!! This CD will format your hard drive, wipe out all data, and install a new CentOS system without asking for your say-so. Tom On Aug 11, 2005, at 6:37 PM, Francesco Peeters wrote: On Fri, August 12, 2005 0:25, Doug Lytle said: Michael Jones wrote: Is there a quick configuration that can be put into place to simply experiment with the system (like create a couple of extensions wit Xlite and make a internal phone system in my office that doesn't really go out anywhere - or maybe connects to a voip provider later)? If you compile from source you can do a make samples . Doug And otherwise, you can try Asterisk @ Home, but be warned that it installs a complete new system, so do not try it on an existing machine you do not wish to be wiped clean first! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list
Re: [Asterisk-Users] TE405P / TE410P with 2nd generation firmware field upgradable?
Matt Florell wrote: Long answer: you have to send it to Digium for them to do an upgrade, they don't have an official process for this yet and won't give you a price, I have called and asked them many times. They also mention upgrades from your 405/410 to a 406/411 are available too, but again no specifics. Supposedly if you have a card with the 2nd gen firmware on it you can upgrade to the third gen firmware, whenever it would come out, in the field. We do 2nd gen firmware upgrades for customers every day, have been doing them for a couple of weeks now. The process is simple, just contact the RMA department and be prepared to pay for shipping the card to and from our facility. Once the card has 2nd gen firmware on it, future upgrades will be possible in place, without having to send the card to us. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P / TE410P with 2nd generation firmware field upgradable?
Andrew Kohlsmith wrote: Unfortunately not. It's a configuration PROM, not EEPROM or FLASH. I am pretty sure that Digium has an upgrade program in place though. It's best to contact them directly for these types of inquiries instead of the list. Actually, it is EEPROM, and will be upgradeable in the field once the card has 2nd gen firmware installed on it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
Well that would explain the choppy/stuttering sound we get on these calls since there is no audio during those error transmissions. According to Kevin's reply I had my timing logic backwards. Should I be using any other timing settings on the Asterisk side?? The tech for our legacy PBX says that the PBX will not provide any timing. One side of the T1 needs to provide timing and the other needs to recover timing from the line. If the PBX will not provide any timing then you need to make sure that Asterisk *is* providing the timing. -Matt -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Sendmail question
RedHat also configures their sendmail to not accept connections from any servers other than the localhost. Although I wouldn't expect that to affect outgoing mail, I have found that it often does. Google for redhat sendmail DAEMON_OPTIONS 127.0.0.1 Or check out this article: http://www.joreybump.com/code/howto/ smtpauth.html and look for DAEMON_OPTIONS . Tom On Aug 12, 2005, at 5:09 AM, Rudolf Ladyzhenskii wrote: Hi, all I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. Using Ethereal, I can see that messages are leaving my network, but receipeint mail server never replies back. As a result, mail delivery is timed out. I got a book on sendmail and it looks quite complex. It will take quite a bit of time to find out what is going on. I am using FC3 and sendmail uses default configuration. Is teher a quick tweak I can do to get it to work? May be someone can suggest another mail program that is easier to setup? Messages sent from command line behave same way as ones sent from asterisks, so it is definetely a sendmail configuration issue. Thanks a lot, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] help on receive text
Hi Matt, Thanks for the information. regards Somesh -Original Message- From: Matt Riddell [mailto:[EMAIL PROTECTED] Sent: Thursday, August 11, 2005 11:43 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] help on receive text someshwarak wrote: Hi * users, I am only seeing SendText in the available asterisk applications. But I have not seen Receive Text application. I tried on asterisk-1.0.7 and 1.0.9. Can anyone tell me how to use this receive text command. The only one that currently exists is the agi command RECEIVE TEXT. I want to use receivetext command and get text information from an softphone so that that can be routed to some other phone supporting text message. (my soft phones are SIP/IAX based). You're going to have to write an AGI for it. where I can get the receive text application for asterisk? You'll find it in the latest CVS HEAD. But it's agi, not an application. pabx*CLI show agi receive text Usage: RECEIVE TEXT timeout Receives a string of text on a channel. Specify timeout to be the maximum time to wait for input in milliseconds, or 0 for infinite. Most channels do not support the reception of text. Returns -1 for failure or 1 for success, and the string in parentheses. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P / TE410P with 2nd generation firmware field upgradable?
On Friday 12 August 2005 09:30, Kevin P. Fleming wrote: Andrew Kohlsmith wrote: Unfortunately not. It's a configuration PROM, not EEPROM or FLASH. I am pretty sure that Digium has an upgrade program in place though. It's best to contact them directly for these types of inquiries instead of the list. Actually, it is EEPROM, and will be upgradeable in the field once the card has 2nd gen firmware installed on it. TE405 Rev A uses an XCF01S which by Xilinx own documentation says is a PROM. I haven't taken a close look at the TE405 Rev B. :-) Of course, Xilinx could be outright lying and the're not PROM at all, or they're being pedantic and saying yes it's programmable, but it's also REprogrammable. Either way I'll just shut up. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] txfax spandsp
I've been using spandsp and rxfax to receive faxes for a while now and over the past few days I've looked into the other side of things - txfax. I can't seem to get it working properly. I've included debug logs below of both the tx and rx side of things. I've tried three different servers, asterisk v1.0.6, 1.0.9 and CVS, FC2 and FC4, libtiff 3.5.7 and 3.7.1, TE405P and TE110P cards, connected to each other via crossover cable (with rxfax as a fax client), the TE405P connected to itself (ports one and two crossover cable), and also connected to the PSTN with a hardfax on the other end. I always get the same errors. Would be very grateful if someone could give me some pointers or otherwise shed some light. The logs below are from spandsp 0.0.2 pre18 on CVS Head txfax talking to 0.0.2 pre18 on v1.0.9 stable rxfax. Doesn't matter though as the last debug entry is always the same (T4 timeout in state 4) * TXFAX Debug * -- Executing TxFAX(Zap/1-1, /tmp/11236686843.tif|caller|debug) in new stack Slow carrier up Urgent handler Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up DIS: 80 00 ce f4 80 80 81 80 80 80 18 DIS with final frame tag In state 10 DIS: Prefer 256 octet blocks Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm 2D coding Scan line length: 215mm Recording length: Unlimited Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm Minimum scan line time for higher resolutions: T15.4 = T7.7 North American Letter (215.9mm x 279.4mm) North American Legal (215.9mm x 355.6mm) DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 20ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Start sending document Start tx document Changed from phase 2 to 4 DCS: 83 00 86 80 80 80 00 HDLC underflow in state 3 Changed from phase 4 to 6 Changed from phase 6 to 3 Slow carrier up T4 timeout in state 4 * End of txfax debug * * RXFAX debug * Aug 12 11:00:12 DEBUG[11937]: pbx.c:1274 pbx_extension_helper: Launching 'RxFAX'-- Executing RxFAX(Zap/1-1, /var/mbox/msgs/1123815612.2.tif|debug) in new stack Urgent handler Aug 12 11:00:12 DEBUG[11937]: channel.c:1752 ast_set_read_format: Set channel Zap/1-1 to read format slin Aug 12 11:00:12 DEBUG[11937]: channel.c:1719 ast_set_write_format: Set channel Zap/1-1 to write format slin Slow carrier up Slow carrier down Aug 12 11:00:12 DEBUG[11937]: chan_zap.c:4138 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Changed from phase 1 to 4 DIS: Prefer 256 octet blocks Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm 2D coding Scan line length: 215mm Recording length: Unlimited Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm Minimum scan line time for higher resolutions: T15.4 = T7.7 North American Letter (215.9mm x 279.4mm) North American Legal (215.9mm x 355.6mm) DIS: 80 00 ce f4 80 80 81 80 80 80 18 Aug 12 11:00:16 DEBUG[11937]: chan_zap.c:4053 zt_read: DTMF digit: f on Zap/1-1 Aug 12 11:00:16 DEBUG[11937]: chan_zap.c:4095 zt_read: Fax already handled HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up DCS: 83 00 86 80 80 80 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 20ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Get at 9600bps, modem 1 Changed from phase 3 to 5 Fast carrier up Fast carrier down Fast carrier up Coarse carrier frequency 1699.90 (68) Training error 0.294910 Training succeeded (constellation mismatch 0.636352) Fast carrier trained Craig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone
It would be nice if there was a dialplan for each registration or line, which would allow me to never press send for any of the systems I register to On my Sipura there isI haven't checked one of the polycoms but I suspect they are no different. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: Thursday, August 11, 2005 8:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone Jonathan k. Creasy wrote: YeahI think that every install I have done the first thing that happens is why is there a delay before the call connects? and the answer is you have to hit dial or wait 10 seconds. What all phones does that apply to? I'm fairly certain it applies to the Polycom phones I've read about, but I'm not sure about others. I'm obviously a newbie to the field as well (well, at least to the physical phones). Actually, I never need to press send when dialling numbers from my polycom phone... well, actually I do, but 99% of people wouldn't... I'll explain: My sip phone registers to different SIP servers (all asterisk), and each server has it's own dialplan, ie, one is purely PSTN calls, and so there is no leading digit for 'external calls', you just dial the number. One is a PBX where you need to dial 9 to get an outside line, with 3 digit extensions. Another is a PBX where you need to dial 0 to get an outside line, with 4 digit extensions. Also, each PBX system has different voicemail numbers to get to voicemailmain etc... So, I don't press send when dialling through my own PBX, but I do if I am dialling through one of the other systems, since their dialplans are not catered for in my polycom. BTW, it would be nice if there was a dialplan for each registration or line, which would allow me to never press send for any of the systems I register to:) Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
On Fri, 12 Aug 2005, Geoff Manning wrote: OK. So I changed it to: span=1,0,0,d4,ami And the Blue Alarms are still occurring but now in conjunction with Slip errors. I feel like I am on the right track though. Which side shows the slips? I am not that familiar with T1, Are you sure the signalling between the pbx and asterisk is set the same on both? Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP301 and 501 with asterisk...
Our vendor told us we can't buy the 841's anymoreanyone else have this problem or have a vendor that is still selling them? -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Ternero Sent: Wednesday, August 10, 2005 9:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP301 and 501 with asterisk... Hi I work with Sipura 841, and works very well, after make a firmware update. Alex -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Trevor Peirce Enviado el: jueves, 11 de agosto de 2005 18:56 Para: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Polycom IP301 and 501 with asterisk... Ing. Marlo R. Beltran G wrote: Hi, I am about to buy ip pbx asterisk system but what ip phones do you recommend? Are polycom ip all functional with the ip pbx system??? We just got a Polycom IP501 for testing and have thus far been unsuccessful at getting it to regiser with asterisk. Outgoing cals work fine now (with authentication; verified with ethereal). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Possibly bad FXS module in TDM400P?
On Fri, Aug 12, 2005 at 08:46:45AM -0400, Jeff Borders wrote: I've got the latest zaptel and cvs asterisk software loaded on my phone server running FC3. And yes, It's fully updated and udev is setup correctly. I've got a TDM400P with one FXS and one FXO module installed. When I load zaptel and wctdm and run ztcfg -vvv, I get this: [EMAIL PROTECTED] ~]# ztcfg -vvv ZT_CHANCONFIG failed on channel 1: No such device or address (6) ztcfg failed to actually perform the ioctl-s on the devices. Is the module loaded? lsmod | grep zaptel ls -l /proc/zaptel/* Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. This is the output from? doesn't ztcfg givr the error above? Regardless of my configuration, it should show up in ztcfg, right? ztcfg first reprorts (when verbose enough) the channels/spans configured in zaptel.conf and only then opens device files. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7960 + 7914 Problems
I'm still having problems getting this to work. I cannot get anything to display on my 7914 other than blank lines. I have SIP/5920-5930 in [main] that I'd like to add to the 7914 and indicate hook status. The 7960 is registering okay as SCCP/5000. What exactly should my sccp.conf file look like? When I make changes to this, how do I enact them? Do I reload Asterisk and reboot the phone or do I have to restart Asterisk and reboot the phone? Right now, this is how sccp.conf looks. [general] keepalive = 5 context = main dateFormat = D-M-Y ; M-D-Y in any order (5 chars max) bindaddr = 192.168.1.10 ; replace 1.2.3.4 with the ip address of the port = 2000; listen on port 2000 (Skinny, default) [SEP00036B75B542] type= 7914 autologin = ,,79140,79141 speeddial = 5929,Craig B,5929 at main [martha] id = 5000 pin = label = Martha context = main callwaiting = 1 mailbox = 5000 ; Another line. [79140] id = 5929 pin = label = Craig context = main [79141] id = 5933 pin = label = Andy context = main Message: 4 Date: Fri, 05 Aug 2005 14:53:39 -0400 From: Joseph [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7914 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain On Fri, 2005-08-05 at 14:09 -0400, Craig Bruenderman wrote: How does one go about programming a Cisco 7914 sidecar to be used as a busy lamp field? In the sccp.conf file, o As a Line: You can assign a line to the button/lamp which is really neat. The lamp is green when you are on the line, blinking green when you put the line on hold, blinks orange when you call that line. If you had a 7960 and wanted a line on the 7914 you could do it this way: autologin = ,,79140,79141 ; This makes it go to the 7th button ; for the first line button. o As a speed dial (lamp is either off or red) You setup a speed dial like this: speeddial = 10,John Doe,[EMAIL PROTECTED] And then to make this work you need to have the exten = 10,hint,SCCP/10 ;sccp phone exten = 10,hint,SIP/10 ;Sip phone As a line you don't need the hint. Note: a speed dial with hint shows an icon of a phone just like a line. And when the lamp is on with the hint, it shows an icon of a phone with an X though it. This is the case with the 7940/7960 speed dials as well. The icon gives you the same line status as the lamps even without the 7914 sidecar. Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firewall will definately increase jitters inyourvoice conversation
On Wed, Aug 10, 2005 at 09:58:09PM -0600, Rich Adamson wrote: That's a crack of crap sold by the marketing (not sales) people selling firewalls. If you know what you're doing, one can very easily secure any linux system to function on the Internet (etc) without a firewall. It all depends on your level of knowledge/skills on how to disable those items that are not really needed in your environment. Start with a 'netstat -a' to identify those ports that are listening, and shut those items down that you don't want exposed. netstat -lutp is more efficient than a simple netstat -a. RTFM netstat. You could also write your own iptables script and optimize it for low latency. In fact, I bet there are enough such scripts rolling around the 'net. Even I wrote one. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P V2 changes?
yes, fedora 3 but without any changes at the sources Master Abi wrote: Are you using Redhat/Fedora? If I remember those init scripts is for Redhat/Fedora. I am using gentoo. Did you make any modifications to wct4xxp.c. or pass any parameters to zaptel. I see there is a #define SUPPORT_GEN1 in to wct4xxp.c which I commented out, but it made no difference. ztcfg seems to where the channels become unassigned. Thanks again. Kib Eki wrote: Hi, we also got one V2 TE405P card. It works fine now. At the moment we use for bridging the Pri to our old PBX. You must use zaptel, libri and asterisk v 1.0.8 or higher. We use 1.0.9 at the moment. zaptel: after make; make install i also executed make config. This copies the correct startup script to /etc/init.d/zaptel. Without this it also didn't worked for me. Master Abi wrote: Hi I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after putting in the upgraded board but did not change any conf, but the spans become active but will not come up. I guess I am missing something or are the any changes to the zaptel/libpri software that is required. I cannot find any info about this or does this new firmware only work with latest CVS. I am using 1.0.9 with 2.6.12 kernel Zapata Telephony Interface Registered on major 196 Found TE4XXP at base address fdfff000, remapped to f8928000 TE4XXP version c01a0164, burst ON, slip debug: OFF TE4XXP running with work queues. FALC version: 0005, Board ID: 00 Reg 0: 0x364e9400 Reg 1: 0x364e9000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a0164 Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE405P (2nd Gen) eth0: link up, 10Mbps, half-duplex, lpa 0x About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Unassigning channel 0/1! Unassigning channel 0/2! Unassigning channel 0/3! Unassigning channel 0/4! Unassigning channel 0/5! Unassigning channel 0/6! Unassigning channel 0/7! Unassigning channel 0/8! Unassigning channel 0/9! Unassigning channel 0/10! Unassigning channel 0/11! Unassigning channel 0/12! Unassigning channel 0/13! Unassigning channel 0/14! Unassigning channel 0/15! Unassigning channel 0/16! Unassigning channel 0/17! Unassigning channel 0/18! etc... This was working for 10 months before the upgrade. Master ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival Problem
I'm attempting to use Festival with Asterisk on an x86_64 system. This IVR application works ok on a P4 system. I'm using the FC3 x86_64 distro on a single processor Opteron system. Festival by itself (using the command line and speakers) seems to work ok, and Asterisk without Festival works ok. When the Asterisk dialplan calls Festival, however, Festival reports a disconnect and Asterisk's Festival command does not complete. Later, when I shut down the system for reboot, I get a kernel panic. I've tried both the FC4 Festival rpm as well as the source download from festvox.org. I modify the siteinit.scm file as per the wiki page, and I use the stock festival.conf file in Asterisk. Has anyone experienced this behavior, and is there a workaround? Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voipjet experiment
Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is causing the problem, or if VoipJet doesn't sense the line was picked up (and thus doesn't pass this info to me). Here is a sample output of CLI when the disconnect happens: --- -- Called voipjet/18004337300 -- Call accepted by 216.118.117.46 (format ulaw) -- Format for call is ulaw -- IAX2/voipjet-1 is making progress passing it to SIP/541-d994 -- Nobody picked up in 3 ms -- Hungup 'IAX2/voipjet-1' --- During the 3 ms I hear the American Airlines auto attendant giving me options, I can choose and option and the auto attendant will recognize the DTMF and send me to that menu, then after a total of 30 seconds, I get disconnected. I haven't had this issue with any other numbers yet (only been in production use one day...) Any info is appreciated. G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Sendmail question
On Fri, Aug 12, 2005 at 07:09:45PM +1000, Rudolf Ladyzhenskii wrote: I want voicemails to be delivered to recepients by e-mail. I have set up voicenail, and I can see asterisk is using sendmail to send messages out. [snip] I got a book on sendmail and it looks quite complex. Right. Asterisk needs a sendmail. A program that provides basically the same command-line interface sendmail provides for sending messages. There are currently many of those. Fedora includes not only Sendmail, but Postfix as well. I personally find Postfix much sainer than Sendmail. Furthermore, what you basically want is a rather simple setup: relay all local messages through a certain SMTP server to some remote locations. There a number of programs, such as ssmtp and nullmailer, that do exactly that, without keeping a local queue at all. (Their error handling is problematic: network errors may mean lost messages: they don't queue. But then again, they're much simpler.) So in short: you have a whole range of programs that provide a sendmil. Not necessarily the complex Sendmail your book was written about. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Ang: [Asterisk-Users] Voipjet experiment
Hej! Jag är på semester vecka 33, åter igen på kontoret 22 aug mvh Gunnar / JMG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
Peter Svensson wrote: Which side shows the slips? The slips are seen on the legacy PBX side (Isoetec 228) I am not that familiar with T1, Are you sure the signalling between the pbx and asterisk is set the same on both? Unfortunately I am not aware of the signalling set on the Isoetec side. Nothing in the console menu's allows me to see it and the tech said he was unsure. Not a good answer. I have tried several signalling settings to no avail. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Weird issues with TDM400P
We have a TDM400P installed here with four FXS modules. It works well except for a couple of issues: First, I have a Panasonic KX-TG2431 telephone (so others can reach me when I am in o ther parts of the building) hooked up to one of the FXS ports. When the other end hangs up, I get the usual CPC disconnect signal. After the CPC, sometimes it will go to a dialtone, and other times a reorder. There doesn't seem to be any rhyme or reason behind what it does, it's random. That in and of itself is not a problem, but it did seem odd. However, sometimes during the reorder (after about 3 seconds) the reorder stops and I get an FSK data burst (assumedly the VMWI.) My understanding is that the data burst isn't supposed to happen until *after* my phone goes on-hook. Any idea what's up here? Second, one of our ports has static on it. A tenant moved in and we configured their fax for our fourth (until then unused) FXS port. They had numerous fax problems, and when I connected a buttset up to their wall jack, there were minor static issues. I swapped their fax connection to a different port on the TDM400P, and the static went away. I put a different line (to elsewhere in the building) into their usual fax port on the TDM400P, and the static showed up there. So, the problem follows the FXS port, and not the line itself. For now, I have given them the FXS port I was using for my Panasonic, and put my Panasonic on the noisy port. It's not all that noticeable, but of course the fax doesn't like any noise at all. Any idea what this could be? I haven't done much troubleshooting yet, but I plan on taking down the * box and re-seating the FXS module. If that doesn't work, I'll swap it and another FXS module on the board around to see if the problem follows the module or the socket. Other than that, any recommendations? Third (minor) issue, which affects all the ports, but I am using my phone as an example: I can call from my desk phone (Polycom IP 501) to my analog phone and answer. Then I can hit flash on the analog phone, and I'll hear the usual double-click on the Polycom as the analog phone flashes. However, between those two clicks, there's some sort of weird noise. It's not a radio station or anything. It's more like a whine of sorts. The same noise can be heard by hooking up a buttset to one of the ports and putting it in monitor mode (where it monitors the line, but doesn't go offhook.) Is it normal for Digium cards to generate this noise, or does this indicate something is wrong with my card? (Our Comdial PBX at church does a similar type of thing when flashing on an analog extension, but that still doesn't tell me if it's normal or not.) Anyone else see this same type of thing? Fourth: I get this message in the log (related to our analog FXS ports) Aug 11 16:46:17 WARNING[23855]: zt hook failed: Device or resource busy I think this may have something to do with getting a dialtone instead of reorder after hangup (the first thing I mentioned.) Not 100% sure though. Anyone have any ideas on any of these? If you can share I'd appreciate it. TIA. Jeremy -- Jeremy Gault[EMAIL PROTECTED] Network Administrator, WinWorld Corporation voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 url: http://www.winworld.cc/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PC for 8 line system
I have 2 TDM04b cards currently running in an asterisk at home box that I am ready to replace with the CVS version of asterisk. What I am looking for is thoughts / recommendations. I want to move this to a small form factor ( shuttle ) machine and was wandering what expeience / advice there was for this? I have seen the incompatible motherboard list at digium ( and in fact I think my current machine is on the list ! ), but wanted to know what others are doing for small form factor tdm setups? Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipjet experiment
Interesting. Something similar for me, except it comes back as busy after about 30 seconds. -- Called [EMAIL PROTECTED]/18004337300 -- Call accepted by 69.25.60.30 (format ulaw) -- Format for call is ulaw -- IAX2/voipjet-1 is making progress passing it to SIP/207-b8f3 -- IAX2/voipjet-1 is making progress passing it to SIP/207-b8f3 -- IAX2/voipjet-1 is busy -- Hungup 'IAX2/voipjet-1' I wonder if American Airlines forgot an Answer in their dialplan ;-) Tim Garth Summey wrote: Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is causing the problem, or if VoipJet doesn't sense the line was picked up (and thus doesn't pass this info to me). Here is a sample output of CLI when the disconnect happens: --- -- Called voipjet/18004337300 -- Call accepted by 216.118.117.46 (format ulaw) -- Format for call is ulaw -- IAX2/voipjet-1 is making progress passing it to SIP/541-d994 -- Nobody picked up in 3 ms -- Hungup 'IAX2/voipjet-1' --- During the 3 ms I hear the American Airlines auto attendant giving me options, I can choose and option and the auto attendant will recognize the DTMF and send me to that menu, then after a total of 30 seconds, I get disconnected. I haven't had this issue with any other numbers yet (only been in production use one day...) Any info is appreciated. G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voipjet experiment
I get the same problem @ home when I use it. I thought it was just me. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth Summey Sent: Friday, August 12, 2005 10:58 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Voipjet experiment Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is causing the problem, or if VoipJet doesn't sense the line was picked up (and thus doesn't pass this info to me). Here is a sample output of CLI when the disconnect happens: --- -- Called voipjet/18004337300 -- Call accepted by 216.118.117.46 (format ulaw) -- Format for call is ulaw -- IAX2/voipjet-1 is making progress passing it to SIP/541-d994 -- Nobody picked up in 3 ms -- Hungup 'IAX2/voipjet-1' --- During the 3 ms I hear the American Airlines auto attendant giving me options, I can choose and option and the auto attendant will recognize the DTMF and send me to that menu, then after a total of 30 seconds, I get disconnected. I haven't had this issue with any other numbers yet (only been in production use one day...) Any info is appreciated. G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip problem
i have configured a sip phone to make calls through a sip server but when i make call through the sip phone to the sip server every thing goes well and the call is done perfectly but on sip server it gives me these messages(i have 2 pc with different ips one with a sip phone and the another with an asterisk ): Aug 12 18:19:11 NOTICE[12149]: chan_sip.c:5284 register_verify: Peer 'wassim' is trying to register, but not configured as host=dynamic Aug 12 18:19:11 NOTICE[12149]: chan_sip.c:8730 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '195.112.214.98' Aug 12 18:19:11 WARNING[12149]: chan_sip.c:7910 handle_response: Forbidden - wrong password on authentication for REGISTER for 'wassim' to '195.112.214.98' Aug 12 18:19:31 NOTICE[12149]: chan_sip.c:4380 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again any body have an idea. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Weird issues with TDM400P
Jeremy Gault wrote: We have a TDM400P installed here with four FXS modules. It works well except for a couple of issues: snip Second, one of our ports has static on it. A tenant moved in and we configured their fax for our fourth (until then unused) FXS port. Confirm it is the module and not the TDM base card by swapping modules and see if the static follows the module, then RAM the module from Digium, assume you can convince them there is a problem. snip However, between those two clicks, there's some sort of weird noise. It's not a radio station or anything. It's more like a whine of sorts. That seems to be a problem in the FXS modules. I have two configured as Ground Start ( which Digium says they don't support ) and when in an idle condition, there is that noise that I presume is some data the module is leaking through. As long as you have a clean talk path I wouldn't be concerned with this minor flaw. Good luck with the rest John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Comedian annoucment files
A user has their unavailable message played and once that message is over the Comedian message is played right after. Is there any way to prevent the Comedian message being played if the user's unavailable/busy message is being played. Thanks, Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] real-time priority , -p switch
On Fri, 2005-08-12 at 08:26 +0200, Peter Svensson wrote: Since Linux is not RTOS, why some folks are using this -p switch? It has no effect on standard Linux box. Linux is not a hard realtime os with guaranteed timing. What the -p flag does is to request the realtime scheduler. This means a process wil no longer be subjected to the stanadrd unix scheduling but rather use a strict priority scheduling. The net result is that once a process using the realtime scheduler is ready to run the kernel wihh schedule it as soon as possible. It will only be preempted by realtime processes of the same or better priority. With the addition of the lowlatency patches the worst case latency for userspace applications is very low. The remaining difference between a hard RT os is the guarantees it can make. Peter Thanks for the explanation, it makes sense now. Though is the way to verify that asterisk is running with -p switch? I've modified the startup script to start asterisk with -p; however asterisk starts several sub-precess-ID's. Do the sup-process-ID's are effected by the -p switch? I run schedtool (12189 is asterisk PID with -p switch) and it shows: schedtool 12189 PID 12189: PRIO 0, POLICY N: SCHED_NORMAL, NICE -15, AFFINITY 0x1 I run schedtool (27421 is asterisk PID without -p switch) and it shows: schedtool 27421 PID 27421: PRIO 0, POLICY N: SCHED_NORMAL, NICE -15, AFFINITY 0x1 I can verify that nice has taken effect but PRIO shows in both cases 0 -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] USB handset wanted
Give me a handset with a ringer and hook that both work under Linux and I will buy 30 pieces on the spot! No kidding.. RESPONSE: We are working on it and want to make sure we have the product definition correct to best serve the Linux community. Please private e-mail me to clarify if your first target application is the Asterisk PBX so there is no misunderstanding regarding your basic requirements. Thanks !!! Best regards...Bill Bill McCready (PCPhoneline.com) wrote: We are planning to develop versions of our USB based phone and gateway products for Linux. The plan is to make them will work like regular phones exactly like our Windows versions do including physically ringing loudly on incoming calls. Which versions of Linux are the most popular at the moment in the workplace so we can decide which one to focus our energies on first? Best regards, Bill McCready PCPhoneline.com - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 11, 2005 1:38 PM Subject: Re: [Asterisk-Users] USB handset wanted Ondrej Valousek wrote: Matt, You have forgotten the ringer. In fact, I don't care that much about LCD buttons. I want to use it with something like X-lite. Initially, I used machine builtin soundcard with X-Lite (worked well) but then I realized that if the phone is supposed to compete with the standard analog phone, it must have a working ringer. Fair enough. From what I see I suppose that every handset with builtin ringer must be recongized to the OS as 2 USB soundcards - one for speaker/mike, the second as a ringer. The ones I have worked with have a seperate ringer that just takes an int to decide which ringtone to play. I.E. it is not shown as a soundcard. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipjet experiment
I seem to remember (calling over isdn) that American Airlines doesn't actually send back a Connect for quite awhile - there's just a -Progress w/in-band info to cause voice cut-through. Or something like that ;) On 8/12/05, Brian C. Fertig [EMAIL PROTECTED] wrote: I get the same problem @ home when I use it. I thought it was just me. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth Summey Sent: Friday, August 12, 2005 10:58 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Voipjet experiment Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is causing the problem, or if VoipJet doesn't sense the line was picked up (and thus doesn't pass this info to me). Here is a sample output of CLI when the disconnect happens: --- -- Called voipjet/18004337300 -- Call accepted by 216.118.117.46 (format ulaw) -- Format for call is ulaw -- IAX2/voipjet-1 is making progress passing it to SIP/541-d994 -- Nobody picked up in 3 ms -- Hungup 'IAX2/voipjet-1' --- During the 3 ms I hear the American Airlines auto attendant giving me options, I can choose and option and the auto attendant will recognize the DTMF and send me to that menu, then after a total of 30 seconds, I get disconnected. I haven't had this issue with any other numbers yet (only been in production use one day...) Any info is appreciated. G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Forwarding behavior question
We use Polycom 501s here and several users utilize the Forward soft-button to forward their extension to another extension or outside to a cell phone when they are out. My question is, how can I configure the dial plan so that if they have forwarded their extension via the phone, and the extension they forwarded to does not answer, return them to the voicemail of the originally dialed extension. E.g. Dial Tom - tom's extension forwarded to Betsy - Betsy's phone rings but she does not answer - return to Tom's voicemail instead of Betsy's. Craig Bruenderman Network Advocates, Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP500 / Registration Question?
Hello again, I have a bunch of Polycom IP500 Phones with Boot 2.6.2 and SIP 1.4.1. I have defined seperate user and peer settings for my extensions as per posts I have seen in here. I can access voicemail...etc and the phone seem work fine. Question: when I do sip show registration there is nothing listed and/or sip show subscriptions nothing is there. But when I do sip show peers I see a list of my phones, same for sip show users. Shouldnt I see my phones as registered or something similar under these two sections? I have them set to register, and like I said they are working fine. Any help? Thanks, Kenny __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] three questions
Hello All,
I just started to use asterisk with Digium card (4 fxo ports)
and I've met some problems ( I'm just new in asterisk so questions may
be stupid )
my environment:
Debian testing,
asterisk 1.0.9
zaptel-1.0.9
TDM04P
1) when asterisk receiving incoming call on TDM card all networking
cards stops to send or receive any data for some time
nothing suspicious in the log files, nothing in dmesg ( even with
highest level of logging)
I was suspected shared interrupts - but ethernet and tdm cards are using
different ones
any ideas ?
2) does someone has simple example config files for using asterisk as a
gateway between SIP/internal extensions and PSTN ?
3) I've set up fax receiving and sending it to email
[fax]
exten = 1,1,Macro(faxreceive)
exten = h,1,System(/var/lib/asterisk/mailfax ${CALLERIDNUM}
${CALLEDFAX} ${EXTNAME} ${EXTMAIL} ${FAXFILE} ${EXTCOMPANY})
but CALLERIDNUM always empty
any suggestions ?
P.S. I'm not expecting answers with full sescriptions if my questoins
are already answered - just give me the link to this page.
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RE: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
Peter Svensson wrote: I am not that familiar with T1, Are you sure the signalling between the pbx and asterisk is set the same on both? I have unearthed some documentation on the programming side of the legacy PBX. I can set the following on the PBX for each line on the T1 card: Line Type: 0 immediate return on signalling 1* wink start, return supervision on answer 2 wink start, return supervision on ring 3 return supervision on answer 4 return supervision on ring I have tried all of those setting and nothing seemed to have any affect. But I feel those settings in conjunction with some settings in zaptel/zapata conf files may help. I am going to try some out, any ideas are welcome! * It is currently set at 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
Geoff Manning wrote: Peter Svensson wrote: I am not that familiar with T1, Are you sure the signalling between the pbx and asterisk is set the same on both? I have unearthed some documentation on the programming side of the legacy PBX. I can set the following on the PBX for each line on the T1 card: Line Type: 0 immediate return on signalling 1* wink start, return supervision on answer 2 wink start, return supervision on ring 3 return supervision on answer 4 return supervision on ring I have tried all of those setting and nothing seemed to have any affect. But I feel those settings in conjunction with some settings in zaptel/zapata conf files may help. I am going to try some out, any ideas are welcome! * It is currently set at 1 Those options are all related to per circuit call signaling. blue alarms refer to timing errors on the T1 i.e. timing loss such that an entire frame is either dropped or repeated. You need to be looking at a lower level ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
Bruce Ferrell wrote: You need to be looking at a lower level Like hw/cabling errors?? If so, that's what I was afraid of for cost reasons. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
Geoff Manning wrote: Bruce Ferrell wrote: You need to be looking at a lower level Like hw/cabling errors?? If so, that's what I was afraid of for cost reasons. Hardware, possible. Unlikely to be cabling. It's usually a timing setting. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
Bruce Ferrell wrote: You need to be looking at a lower level Like hw/cabling errors?? If so, that's what I was afraid of for cost reasons. no - the stuff you found relates to configuring one 64k channel of the T1, you need to find the settings to configure the overall t1. - someone mentioned the encoding settings - its not that or you would be getting more than just a blue alarm, it would not be working period. - you need to look for clock source settings, and set those. have your asterisk t1 card generate the clock on the asterisk TX side for sure. it sounds like the pbx just generates its own tx clock based on the rx data which is what you want (that might be faulty who knows). if it can generate its own tx clock from a crystal try that instead and see if it helps. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
Geoff Manning wrote: Bruce Ferrell wrote: You need to be looking at a lower level Like hw/cabling errors?? If so, that's what I was afraid of for cost reasons. just out of curiousity - what are you paying for a T1 cable that you are worried about cost ? you do realize any old ethernet cable will work right ? we are only talking about 1/8th to 1/80th of the speed. Hardware, possible. Unlikely to be cabling. It's usually a timing setting. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comedian annoucment files
http://www.voip-info.org/tiki-index.php?page=Asterisk+CMD+voicemail On Fri, 2005-08-12 at 11:37 -0400, kurt x wrote: A user has their unavailable message played and once that message is over the Comedian message is played right after. Is there any way to prevent the Comedian message being played if the user's unavailable/busy message is being played. Thanks, Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
Jon Pounder wrote: Geoff Manning wrote: Bruce Ferrell wrote: You need to be looking at a lower level Like hw/cabling errors?? If so, that's what I was afraid of for cost reasons. just out of curiousity - what are you paying for a T1 cable that you are worried about cost ? you do realize any old ethernet cable will work right ? we are only talking about 1/8th to 1/80th of the speed. What? Do you think nickels grow on trees? ;) I was concerned more about the pricing of a T1 *card*, Cabling I'm not concerned with. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
