Re: [asterisk-users] wct4xxp Interrupts not detected with dahdi 2.6, but working ok with 2.5
On Mar 22, 2012, at 11:25 PM, Shaun Ruffell wrote: On Fri, Mar 16, 2012 at 11:13:14PM +0400, Vahan Yerkanian wrote: I've tried upgrading one of my servers with yum update to the latest dahdi/asterisk, and found out that my 4th gen TE410P is failing the dahdi init with Running dahdi_cfg: DAHDI startup failed: Input/output error Rolling back to 2.5 restores the normal operation, and reading the dahdi 2.6 change log I think I'm hitting this bug fix with my mobo/card combo? Vahan, Just closing out this public thread. dahdi-linux 2.6.1 will contain what fixed this issue on your machine. I committed onto both trunk [1] and onto the 2.6 branch [2]. [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10559 [2] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10565 Thanks again for your help, SHaun I'd like to publicly thank Shaun for the level of support he provided for the resolution of this issue. Within 10 minutes of my original email he replied in private asking for the ssh access and spent over 4 hours overall just on my machine, over weekend and after-hours, going through the source revisions and eventually finding the problem. I can only wish that all telecom vendors were like this… Thanks Shaun! Best regards, Vahan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] wct4xxp Interrupts not detected with dahdi 2.6, but working ok with 2.5
Hi, I've tried upgrading one of my servers with yum update to the latest dahdi/asterisk, and found out that my 4th gen TE410P is failing the dahdi init with Running dahdi_cfg: DAHDI startup failed: Input/output error Rolling back to 2.5 restores the normal operation, and reading the dahdi 2.6 change log I think I'm hitting this bug fix with my mobo/card combo? 2011-12-14 19:02 + [r10379-10380] Shaun Ruffell sruff...@digium.com With dahdi 2.6 I'm getting this: #cat /proc/interrupts 209: 1 0 IO-APIC-level wct4xxp No interrupts?! #dmesg kernel: ACPI: PCI Interrupt :02:01.0[A] - GSI 24 (level, low) - IRQ 209 kernel: wct4xxp :02:01.0: Firmware Version: c01a016c kernel: wct4xxp :02:01.0: FALC Framer Version: 2.1 or earlier kernel: wct4xxp :02:01.0: Found a Wildcard: Wildcard TE410P (4th Gen) kernel: VPM450: echo cancellation for 128 channels kernel: wct4xxp :02:01.0: VPM450: hardware DTMF disabled. kernel: wct4xxp :02:01.0: VPM450: Present and operational servicing 4 span(s) kernel: wct4xxp :02:01.0: TE4XXP: Span 1 configured for CCS/HDB3/CRC4 kernel: wct4xxp :02:01.0: RCLK source set to span 1 kernel: wct4xxp :02:01.0: System timing mode, RCLK set to span 1 kernel: wct4xxp :02:01.0: TE4XXP: Span 2 configured for CCS/HDB3/CRC4 kernel: wct4xxp :02:01.0: RCLK source set to span 1 kernel: wct4xxp :02:01.0: System timing mode, RCLK set to span 1 kernel: wct4xxp :02:01.0: TE4XXP: Span 3 configured for CCS/HDB3/CRC4 kernel: wct4xxp :02:01.0: RCLK source set to span 3 kernel: wct4xxp :02:01.0: Recovered timing mode, RCLK set to span 3 kernel: wct4xxp :02:01.0: SPAN 3: Primary Sync Source kernel: wct4xxp :02:01.0: Interrupts not detected. With dahdi 2.5 everything is OK: #cat /proc/interrupts 201: 9157 962863 IO-APIC-level wct4xxp #dmesg kernel: ACPI: PCI Interrupt :02:01.0[A] - GSI 24 (level, low) - IRQ 201 kernel: wct4xxp :02:01.0: Found TE4XXP at base address f200, remapped to f887c000 kernel: wct4xxp :02:01.0: Firmware Version: c01a016c kernel: wct4xxp :02:01.0: Burst Mode: On kernel: wct4xxp :02:01.0: Octasic Optimizations: Enabled kernel: wct4xxp :02:01.0: FALC Framer Version: 2.1 or earlier kernel: wct4xxp :02:01.0: Board ID: 00 kernel: wct4xxp :02:01.0: Reg 0: 0x37554400 kernel: wct4xxp :02:01.0: Reg 1: 0x37554000 kernel: wct4xxp :02:01.0: Reg 2: 0x kernel: wct4xxp :02:01.0: Reg 3: 0x kernel: wct4xxp :02:01.0: Reg 4: 0x3101 kernel: wct4xxp :02:01.0: Reg 5: 0x kernel: wct4xxp :02:01.0: Reg 6: 0xc01a016c kernel: wct4xxp :02:01.0: Reg 7: 0x1f00 kernel: wct4xxp :02:01.0: Reg 8: 0x kernel: wct4xxp :02:01.0: Reg 9: 0x00ff0031 kernel: wct4xxp :02:01.0: Reg 10: 0x004a kernel: wct4xxp :02:01.0: Found a Wildcard: Wildcard TE410P (4th Gen) [snip] wct4xxp :02:01.0: TE4XXP: Span 1 configured for CCS/HDB3/CRC4 wct4xxp :02:01.0: 2G: Got interrupt, status = 010c, CIS = 0080 wct4xxp :02:01.0: RCLK source set to span 1 wct4xxp :02:01.0: System timing mode, RCLK set to span 1 wct4xxp :02:01.0: TE4XXP: Span 2 configured for CCS/HDB3/CRC4 wct4xxp :02:01.0: 2G: Got interrupt, status = 010c, CIS = 0080 wct4xxp :02:01.0: RCLK source set to span 1 wct4xxp :02:01.0: System timing mode, RCLK set to span 1 wct4xxp :02:01.0: TE4XXP: Span 3 configured for CCS/HDB3/CRC4 wct4xxp :02:01.0: 2G: Got interrupt, status = 010d, CIS = 0081 wct4xxp :02:01.0: RCLK source set to span 3 wct4xxp :02:01.0: Recovered timing mode, RCLK set to span 3 wct4xxp :02:01.0: 2G: Got interrupt, status = 010a, CIS = 0080 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: SPAN 3: Primary Sync Source wct4xxp :02:01.0: TE4XXP: Span 4 configured for CCS/HDB3/CRC4 wct4xxp :02:01.0: 2G: Got interrupt, status = 000d, CIS = 0084 wct4xxp :02:01.0: RCLK source set to span 3 wct4xxp :02:01.0: Recovered timing mode, RCLK set to span 3 wct4xxp :02:01.0: 2G: Got interrupt, status = 000b, CIS = 0088 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000b, CIS = 008a wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000a, CIS = 0080 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000b, CIS = 0085 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000a, CIS = 0080 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000b, CIS = 008a wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000a, CIS = 0080 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got interrupt, status = 000a, CIS = 0080 wct4xxp :02:01.0: Reg 5 is wct4xxp :02:01.0: 2G: Got
Re: [asterisk-users] Linksys/Cisco 504G randomly restarts
On 8/15/11 10:46 PM, C F wrote: I have 3 Linksys/Cisco 504G phones they keep restarting at what seems to be random. Sometimes as short as 6 minutes. FW version is 7.4.3a I have searched and tried disabling FW check and all related settings. I also extended all the default 3600 resync checks to a lot longer. TIA CF Hi, Try upgrading to the latest version. I have tens of 504G operating without any problems. How are you powering these phones? I had a case when a PoE switch was experiencing short-circuit problems on a badly wired cable, and was unable to provide enough current to power the phones on the other ports. Replacing the faulty cable fixed the problem. You can always try to power the phones using 5volts DC, 2A center pin positive power source and see if the problem persists. Also I have a Linksys SPA-941 that has a public IP and reboots itself whenever someone tries to bruteforce into it by sending tons of sip registers :) HTH, Vahan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
On 7/30/11 7:39 AM, Bruce B wrote: I think this should be a quick fix since it's rendering the latest stable version useless and making the impression that it was released just to break things and force people onto 1.8x. Just a thought...no blame game. But really something like this should be tackled quickly. No point to break things so badly on the last stable version. Regards, Confirming this issue on 10+ 1.6.2.19 Asterisk servers. This problem makes it difficult to do edits on sip.conf on production systems, as there is ~25% chance that you'll crash the server and cut the established calls. The problem does not exist in 1.6.2.18... I think this problem should be fixed or the 1.6.2.19 should be removed from the digium repo. Regards, Vahan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?
On 5/6/11 11:52 PM, Andrew Latham wrote: On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanianva...@arminco.com wrote: Has anyone used this board as an Asterisk server? http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=HIPMI=Y I'm mostly interested about the possible compatibility issues this board may have with the AEX800 card. Yes that is a great system and the built-in IPMI is a livesaver... if you are using a full size harddrive you need to apply some protection to the card in the case (the superserver 1U). They are close but not touching... Thanks for the info, are you using the AEX800 with it? How's the load, and what actual performance do you have? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?
Has anyone used this board as an Asterisk server? http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=HIPMI=Y I'm mostly interested about the possible compatibility issues this board may have with the AEX800 card. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Safe to upgrade to Centos 5.6 now ???
On 4/14/11 5:03 PM, m...@tdiehl.org wrote: Yes, I noticed that also. For some reason the latest Dahdi rpms are sitting in the top level dir at http://packages.asterisk.org/centos/5/current/ but they are not signed. They need to be signed and moved into the approiate arch directory and the yum metadata rebuilt for them to be seen by yum. FYI, the issues have been resolved, everything installs clearly now. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Safe to upgrade to Centos 5.6 now ???
On 4/14/11 5:03 PM, m...@tdiehl.org wrote: On Thu, 14 Apr 2011, Vahan Yerkanian wrote: A word of notice: asterisk/digium yum repos xmls haven't been updated yet (properly): Yes, I noticed that also. For some reason the latest Dahdi rpms are sitting in the top level dir at http://packages.asterisk.org/centos/5/current/ but they are not signed. They need to be signed and moved into the approiate arch directory and the yum metadata rebuilt for them to be seen by yum. Can we get this fixed properly please? Currently it's giving CentOS 5.6 kernel dependency error due to wrong package name: # uname -r 2.6.18-238.5.1.el5 # yum list installed kernel* Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile Installed Packages kernel.x86_64 2.6.18-238.5.1.el5 installed kernel-headers.x86_64 2.6.18-238.5.1.el5 installed # yum install asterisk16-dahdi Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile Setting up Install Process Resolving Dependencies -- Running transaction check --- Package asterisk16-dahdi.x86_64 0:1.6.2.17.2-1_centos5 set to be updated -- Processing Dependency: libopenr2 for package: asterisk16-dahdi -- Processing Dependency: dahdi-linux for package: asterisk16-dahdi -- Processing Dependency: dahdi-linux-kmod for package: asterisk16-dahdi -- Running transaction check --- Package dahdi-linux.x86_64 0:2.4.1.2-1_centos5 set to be updated -- Processing Dependency: dahdi-firmware for package: dahdi-linux --- Package kmod-dahdi-linux.x86_64 0:2.4.1.2-1_centos5.2.6.18_238.9.1.el5 set to be installed -- Processing Dependency: kernel-x86_64 = 2.6.18-238.9.1.el5 for package: kmod-dahdi-linux --- Package libopenr2.x86_64 0:1.2.0-1_centos5 set to be updated -- Running transaction check --- Package dahdi-firmware.noarch 0:2.0.2-1_centos5 set to be updated -- Processing Dependency: dahdi-linux-fwload-vpmadt032-kmod for package: dahdi-firmware --- Package kmod-dahdi-linux.x86_64 0:2.4.1.2-1_centos5.2.6.18_238.9.1.el5 set to be installed -- Processing Dependency: kernel-x86_64 = 2.6.18-238.9.1.el5 for package: kmod-dahdi-linux -- Running transaction check --- Package kmod-dahdi-linux.x86_64 0:2.4.1.2-1_centos5.2.6.18_238.9.1.el5 set to be installed -- Processing Dependency: kernel-x86_64 = 2.6.18-238.9.1.el5 for package: kmod-dahdi-linux --- Package kmod-dahdi-linux-fwload-vpmadt032.x86_64 0:2.4.1.2-1_centos5.2.6.18_238.9.1.el5 set to be installed -- Processing Dependency: kernel-x86_64 = 2.6.18-238.9.1.el5 for package: kmod-dahdi-linux-fwload-vpmadt032 -- Finished Dependency Resolution kmod-dahdi-linux-2.4.1.2-1_centos5.2.6.18_238.9.1.el5.x86_64 from asterisk-current has depsolving problems -- Missing Dependency: kernel-x86_64 = 2.6.18-238.9.1.el5 is needed by package kmod-dahdi-linux-2.4.1.2-1_centos5.2.6.18_238.9.1.el5.x86_64 (asterisk-current) kmod-dahdi-linux-fwload-vpmadt032-2.4.1.2-1_centos5.2.6.18_238.9.1.el5.x86_64 from digium-current has depsolving problems -- Missing Dependency: kernel-x86_64 = 2.6.18-238.9.1.el5 is needed by package kmod-dahdi-linux-fwload-vpmadt032-2.4.1.2-1_centos5.2.6.18_238.9.1.el5.x86_64 (digium-current) Error: Missing Dependency: kernel-x86_64 = 2.6.18-238.9.1.el5 is needed by package kmod-dahdi-linux-2.4.1.2-1_centos5.2.6.18_238.9.1.el5.x86_64 (asterisk-current) Error: Missing Dependency: kernel-x86_64 = 2.6.18-238.9.1.el5 is needed by package kmod-dahdi-linux-fwload-vpmadt032-2.4.1.2-1_centos5.2.6.18_238.9.1.el5.x86_64 (digium-current) You could try using --skip-broken to work around the problem You could try running: package-cleanup --problems package-cleanup --dupes rpm -Va --nofiles --nodigest I believe for the x86_64 the kernel package name should be kernel.x86_64, not kernel-x86_64: # yum list kernel-* Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile Installed Packages kernel.x86_64 2.6.18-238.5.1.el5installed kernel-headers.x86_64 2.6.18-238.5.1.el5installed Available Packages kernel-debug.x86_64 2.6.18-238.5.1.el5updates kernel-debug-devel.x86_64 2.6.18-238.5.1.el5updates kernel-devel.x86_64 2.6.18-238.5.1.el5updates kernel-doc.noarch 2.6.18-238.5.1.el5updates kernel-xen.x86_64
Re: [asterisk-users] Safe to upgrade to Centos 5.6 now ???
On 4/14/11 1:04 AM, Shaun Ruffell wrote: On Wed, Apr 13, 2011 at 01:00:36PM -0700, Jian Gao wrote: Centos 5.6 came out. Any one tried to update to the 5.6 yet? I am running Asterisk 1.8 and is there any risk to upgrade to Centos 5.6? I'm not sure about Asterisk in general, but if you use DAHDI, please be sure to install version 2.4.1.2. http://lists.digium.com/pipermail/asterisk-announce/2011-April/000313.html A word of notice: asterisk/digium yum repos xmls haven't been updated yet (properly): # yum clean all yum list dahdi* kmod* [snip] dahdi-firmware.noarch 2.0.2-1_centos5digium-current dahdi-firmware-hx8.noarch 2.06-1_centos5 digium-current dahdi-firmware-oct6114-064.noarch 1.05.01-1_centos5 digium-current dahdi-firmware-oct6114-128.noarch 1.05.01-1_centos5 digium-current dahdi-firmware-tc400m.noarch MR6.12-1_centos5 digium-current dahdi-firmware-vpmadt032.noarch 1.07-1_centos5 digium-current dahdi-linux.x86_64 2.4.1-1_centos5asterisk-current dahdi-linux-devel.x86_64 2.4.1-1_centos5asterisk-current dahdi-linux-hpec-kmod-base.x86_64 2.1.0.4-1_centos5.2.6.18_128.1.16.el5 digium-current dahdi-tools.x86_64 2.4.0-2_centos5asterisk-current dahdi-tools-doc.x86_64 2.4.0-2_centos5asterisk-current dahdihpec_enable.x86_64 1.0-1_centos5 digium-current kmod-dahdi-linux.x86_64 2.4.1-1_centos5.2.6.18_194.32.1.el5asterisk-current kmod-dahdi-linux-fwload-vpmadt032.x86_64 2.4.1-1_centos5.2.6.18_194.32.1.el5digium-current kmod-dahdi-linux-fwload-vpmadt032-xen.x86_64 2.4.1-1_centos5.2.6.18_194.32.1.el5digium-current kmod-dahdi-linux-hpec.x86_64 2.4.1-1_centos5.2.6.18_194.32.1.el5digium-current kmod-dahdi-linux-hpec-xen.x86_64 2.4.1-1_centos5.2.6.18_194.32.1.el5digium-current kmod-dahdi-linux-xen.x86_64 2.4.1-1_centos5.2.6.18_194.32.1.el5asterisk-current The yum isn't picking the RPMs for 2.6.18-238.5.1.el5 from http://packages.digium.com/centos/5/current/ and http://packages.asterisk.org/centos/5/current/ So if you just yum update everything to 5.6, DAHDI will fail to load as there are no modules for 2.6.18-238.5.1.el5, as reported by repo xmls. HTH, Vahan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
On 9/27/10 8:57 PM, Michelle Dupuis wrote: HAAST runs a sync script a regular intervals (time to sync data prior to a failover check etc) HAAST includes a sample script which syncs voicemail (and config, etc) files using rsync from master to slave. After a master/slave reversal the process automatically reverses. MD What about ODBC/IMAP voicemail storage? Works great with MySQL MasterMaster replication for me. Vahan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing for residential voip usage
On 6/17/10 12:49 AM, Steve Edwards wrote: On Wed, 16 Jun 2010, Landy Landy wrote: I'm unable to place any calls through a2billing. I followed instructions here: http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/F.A.Q to DISABLE PIN number request Prompt for some users but, I'm not able to place any calls. I created a trunk with the same name as in my sip.conf and I'm not able to make any calls. I don't know what I'm missing. This is the output when trying to call: == Using SIP RTP CoS mark 5 -- Executing [812022418...@a2billing:1] Answer(SIP/1433631307-0015, ) in new stack -- Executing [812022418...@a2billing:2] Wait(SIP/1433631307-0015, 2) in new stack -- Executing [812022418...@a2billing:3] AGI(SIP/1433631307-0015, a2billing.php,3) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php --SIP/1433631307-0015AGI Script a2billing.php completed, returning -1 I can't debug it or anything I'm stuck please help. If you have CLI version of PHP installed, you can also try running /var/lib/asterisk/agi-bin/a2billing.php directly from the shell, and keep feeding it CR/LF, you'll see step-by-step variable assignment and hopefully the error message that stops it from working. You'll need display_errors on in php.ini for this as well. Most probably you're missing a PHP module or your SQL connection is failing. HTH, Vahan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SS7 over an FXO interface
On 4/16/10 3:15 PM, mosbah abdelkader wrote: Hello, Is it possible to transfer ss7 signaling over an FXO interface. I need to setup an ss7 test system composed by two Asterisk based IP-PBX systems with anlog interfaces only (FXO and FXS). I want to know if it is possible to connect the two IP-PBX as following: - FXS interface in PBX1 - connected to - FXO interface in PBX2 = used to transport ss7 signaling. - FXS interface in PBX2 - connected to - FXO interface in PBX1 = used to transport voice between the two PBXs. This connection can be replaced by a simple SIP trunk. Is this scenario possible with libss7 and asterisk. If yes, please give some instructions and tips. Thanks. NO, you can't pass signaling over an analog FXO. You can pass SS7 over Ethernet or E1/T1 links, least to mention. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Converting GSM calls to SIP
On 4/15/10 1:26 AM, Tonty T wrote: That's is all the overhead I am trying to avoid. What I need is a DID with unlimited channel, but they do not offer DIDs in that country. I wanted to know for example when I get a DID from lets say Vitelity, with unlimited channel, what are they using to forward the calls via SIP or IAX to my server? If I knew the details of the process, I could probably tell them to used this method and route the short code to me via SIP. And if it requires hardware I could invest in it myself and have them host it. If their switch doesn't support SIP or doesn't have SIP module installed, there isn't much you can do to get traffic in pure SIP form. Ask them if they can and willing to serve you the traffic via multiple E3 or even better, STM fiber links. STM over fiber is the cheapest way to transport that much channels by means of cabling - you just need 2 strands for TX/RX or even 1 strand if you go with WDM. However the carrier crade hardware for it is *very expensive*. On your side you demux STM link(s) into E3/E1s using expensive carrier grade equipment like Cisco's $25k+ (used) STM cards for Cisco 7500 and up models or if you're smart enough to know where to dig, dirt cheap (~$2K for STM-1 to 24E1) Taiwanese/Chinese media converters. Oh and yes, this isn't a task for a single Asterisk server. The most I've seen a single box capable of is 16 E1s (2 x 8E1 cards) in a single chassis doing only G711a to SIP conversion. HTH, Vahan On Wed, Apr 14, 2010 at 2:46 PM, Jeff Brower jbro...@signalogic.com mailto:jbro...@signalogic.com wrote: On Wed, Apr 14, 2010 at 10:33 AM, Tonty T ton...@gmail.com mailto:ton...@gmail.com wrote: This is a solution they proposed, using GSM gateways, but it wont let me handle 1000 simultaneous calls, the other option was using an E1 but the cost would be too much to deploy 35 E1s to support that many calls. There might be a better way of doing it. If you are planning on having 1000 simultaneous calls, you're going to be looking at a hefty price tag one way or the other. Things to consider - if you're going to have 1000 concurrent calls going out over VoIP trunks (SIP / IAX / whatever), you need to have enough bandwidth to comfortably handle that many calls (each g729 is 8Kb/s bandwidth (but you need to pay a license fee for each channel of g729), each g711alaw is 64Kb/s, etc). That amount of bandwidth won't be cheap, plus the cost of the ITSP giving your 1000 concurrent channels to call on. On the other hand, if you have a bank of E1's, which support (I think) at max 30 concurrent voice channels, you'd need 34 available E1 spans. I'm not sure if you can get 34 spans working in a single asterisk server (there was some discussion about this recently on this list), and you'd have the cost of 34 E1 spans as well. All good points. It might be worth mentioning that including IP/UDP/RTP packet overhead, actual bandwidth is 40 kbps for G729 and 96 kbps for G711. -Jeff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Problems with Linksys IP Phone SPA 942
On 2/25/10 6:50 AM, Tilghman Lesher wrote:
DHCP is designed in such a way that you can legitimately have multiple DHCP
servers on the same network. The first DHCP server which replies and meets
the DHCP client's requirements will be the server to which the client
registers. If the Linksys DHCP server is faster (or if you have several
switches and it replies to some hosts faster), then those hosts will likely
use the Linksys as their DHCP server.
You could technically avoid this situation by provisioning some DHCP option
that the Linksys does not and making all of your DHCP clients require that
option, but that takes quite a bit away from the zeroconf usage of DHCP.
Or you could set up a rule on your managed switch such that broadcasts to
UDP port 67 only hit the switch port on which your intended DHCP server is
located.
I've been successfully using the following to catch Linksys phones and
provide DHCP services only to them for the past few years:
class Linksys {
match if ( substring (hardware,1,3) = 00:0e:08 ) or (
substring (option vendor-class-identifier, 0, 7) = LINKSYS );
}
pool {
allow members of Linksys;
deny dynamic bootp clients;
range 10.168.172.100 10.168.172.250;
}
HTH,
Vahan
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Re: [asterisk-users] Looking for a patch cable for my SPA941 Phones
Wolfgang Pichler wrote: Or can anyone here tell me where to get good (and not to expensive) 2.5mm plug connection binaural headsets ? Ebay might be a source for these: http://shop.ebay.com/items/?_nkw=2.5mm+to+3.5mm+adapter+headphone ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Country numbering plan resources
I find the idea very interesting and quite useful. There is an ongoing effort on Wikipedia to gather as much information as possible, and keep it current: http://en.wikipedia.org/wiki/List_of_country_calling_codes Here is the data for Armenia (phone numbers are 11 digits): http://en.wikipedia.org/wiki/%2B374 regards, Vahan Alex Balashov wrote: On Fri, December 12, 2008 10:57 pm, Michael wrote: So therefore the over all USA and NA % is smaller from this part of the world, hence the up line can make enough profit over all that they are less likely to view it as a loosing proposition. That depends entirely on who your users are calling in North America. I know a customer that got a nice blended deal from a Tier 1 NA carrier for terminating traffic from overseas. Said carrier is pulling their hair out trying to figure out how to get rid of this contract; the customer is cherry-picking the most expensive routes off that plan precisely because it is blended. It's costing them hundreds of thousands of dollars a month. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bug in Asterisk 1.4.22?
Abel Monzon wrote: and then in my softphone I call to 1 the asterisk log say this: -- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php == a2billing.php: Failed to execute '/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory -- Executing [EMAIL PROTECTED]:4] Wait(SIP/abel-28c18000, 2) in new stack == Spawn extension (default, 1, 4) exited non-zero on 'SIP/abel-28c18000' Abel, While this is not an a2billing mailing list and you should get more help in their forum, my guess is that the path to the php cli executable is incorrect in the a2billing.php. It's in the first line of it. HTH, Vahan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Generating 484 Address Incomplete
Hi, We are processing lots of calls and I want to filter these that have incomplete numbers sent with a proper SIP response. These numbers are not in the local dialplan by themselves, so I'm trying to find a way to generate 484 Address Incomplete SIP response based on the length of the extension called. Congestion response is too lossy of the original cause and doesn't work in my case. Any ideas? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Write Asterisk CDR MySQL records to multiple servers
Ricardo Melendez wrote: Hi to all, I actually have an asterisk server configured to write CDR mysql records in the same machine (localhost), but I want to write this records to another machine also in mysql at the same time, It is possible? It means that I want save the records in both machines. One way of doing is this is to setup the second machine as a MySQL slave for your current machine. You can specify which MySQL databases/tables you want to be mirrored, you can find lots of tutorials how to do this on the Web. HTH, Vahan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hint() extension in AEL
Hi,
I've been trying to setup hinting recently on 1.4.20.1, and was
wondering if there is a more elegant way to do the following
piece of dialplan without repeating the hints for every existing
extension/user?
context Main {
hint(SIP/10301) 10301 = call(${EXTEN});
hint(SIP/10301) 301 = call(10${EXTEN});
// [snip]
hint(SIP/10327) 10327 = call(${EXTEN});
hint(SIP/10327) 327 = call(10${EXTEN});
_3XX= call(10${EXTEN});
_103XX =call(${EXTEN});
}
macro call( ext ) {
Dial(SIP/${ext},20,otL(360:61000:3));
switch(${DIALSTATUS}) {
case BUSY:
Voicemail([EMAIL PROTECTED],b);
break;
default:
Voicemail([EMAIL PROTECTED],u);
};
catch a {
VoiceMailMain([EMAIL PROTECTED]);
return;
};
Hangup;
};
So far I tried having the sip extension to ' = jump _103XX' or simply
'= Noop()'. Neither worked,
latter just overwriting the _103XX extension and causing just noop and
hangup executed when you call that extensions.
Any ideas?
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Re: [asterisk-users] [FreeBSD 6.3] Why not use safe_asterisk?
Vincent wrote: Hello I'm running Asterisk 1.4.20.1 on a FreeBSD 6.3 host, and unless I'm mistaken, it seems like /usr/local/etc/rc.d/asterisk script doesn't make use of /usr/local/sbin/safe_asterisk to restart Asterisk in case it crashes. Is this correct, and if yes, why not use it? Thank you. I've been wondering about that myself for a while too :) MySQL is known to be using that method under FreeBSD for quite some time similarly, by running the the /usr/local/bin/mysqld_safe from /usr/local/etc/rc.d/mysql-server. Any active contributors to the net/asterisk wanna shed some light on this mystery? Vahan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
Andrew Kohlsmith wrote: On Saturday 29 September 2007 18:43:59 Andrew Joakimsen wrote: That's horrible. I don't buy too many IP phones these days, but can anyone suggest a place better than the scumbags at VoIP supply? I don't know about you, but I've had nothing but very good results with VOIPSupply. I didnt do huge business with them, but I have purchased new and refurb polycoms from them without so much as an ounce of pain. -A. I've bought more than $10k worth of equipment from voipsupply.com across the globe and they've always treated me very professionally. All their shipments always arrived on time and were well packed and documented. Just my 2 cents, Vahan ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral's Allison is having trouble speaking clearly
Todd Reese wrote: Hi all, I have just install and licensed Cepstral's Allison08kHz on my Asterisk 1.4.11 system. I can call the Allison's extension from my Grandstream IP Phone and she's clear as a bell, but when a call to her extension traverses through one of the Linksys/Sipura 3102 or 2002, she's got the jitters bad. The SPA-202 has only an extension phone on it and the SPA-3102 is my FXO from my Vonage Motorola box. Any clues where to start looking to clear this up? Make sure that you have RTP Packet Size changed to 0.02 from the default 0.03 in the Sipura SIP tab. This is known to cause jitter with Asterisk. HTH, Vahan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Peers show command
Ronaldo Z. Afonso wrote: Hi all, What does (T) mean on the output of iax2 show peers? The following my output. darkstar*CLI iax2 show peers Name/UsernameHost Mask Port Status ronaldo (Unspecified) (D) 255.255.255.255 0 UNKNOWN sp/ata201.26.67.102 (S) 255.255.255.255 4569 (T) UNKNOWN 2 iax2 peers [0 online, 2 offline, 0 unmonitored] Ronaldo. FYI, (T) stands for Trunked peer. Basically means that the communication with that particular host is optimal, with all of the channels using the same packet envelope. HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-942 TFTP Provisioning
Jeremiah Millay wrote: I'm trying to provision some spa-942 phones via TFTP. The phones get their address from a dhcp server which sends it option 66 (address of the tftp server). After spending some time with the phones and even breaking down to sniff traffic from the phones I see that they are not requesting their config from tftp. I can kind of fake the phones into grabbing their configs by doing something like: Make sure you reset to factory default those phones. Quite possible you've disabled resync on reboot or something like that. Our SPA-941 are resyncing from dhcp ok. HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk core dumps on a Sipura forwarded to a queue/moh
Greetings all,
I'm running Asterisk 1.2.9.1 installed from /usr/ports/net/asterisk on
FreeBSD 6.1-RELEASE.
I'm experiencing a guaranteed asterisk core dump with any Sipura device
set to forward all calls to an extension that is mapped to a queue:
-- Executing Macro(SIP/10040-4c43, call|10027) in new stack
-- Executing Set(SIP/10040-4c43, ext=10027) in new stack
-- Executing Dial(SIP/10040-4c43, SIP/10027|20|o) in new stack
-- Called 10027
-- Got SIP response 302 Moved Temporarily back from 10.20.30.40
-- Now forwarding SIP/10040-4c43 to 'Local/[EMAIL PROTECTED]' (thanks to
SIP/10027-4f37)
sip*CLI
Disconnected from Asterisk server
#
so the 10027 is the Sipura-3000 in this case, with configured Cfwd All
Dest: (forward all calls) to the extension 111, which is a queue or
109, which is a musiconhold call.
-rw--- 1 root wheel 11292672 Jul 19 21:15 asterisk.core
(gdb) bt
#0 0x2829d4ab in pthread_testcancel () from /usr/lib/libpthread.so.2
#1 0x28295e3c in pthread_mutexattr_init () from /usr/lib/libpthread.so.2
#2 0x2810a450 in ?? ()
(gdb) bt full
#0 0x2829d4ab in pthread_testcancel () from /usr/lib/libpthread.so.2
No symbol table info available.
#1 0x28295e3c in pthread_mutexattr_init () from /usr/lib/libpthread.so.2
No symbol table info available.
#2 0x2810a450 in ?? ()
No symbol table info available.
If I set the Cfwd All Dest: in the Sipura configuration interface to a
phone extension (f.e. 10011) everything works ok.
Any clue what's causing this?
--8-- extensions.ael ---8--
context default {
s =Goto(MainIVR|s|1);
};
context Main {
includes {
Gateways;
MainENUM;
};
s =Goto(MainIVR|s|1);
101 = Queue(InfoDesk);
111 = Queue(Support);
121 = Queue(Accounting);
131 = Queue(Admin);
141 = Queue(DomHosting);
};
--8-- extensions.ael ---8--
--8-- queues.conf ---8--
[Support]
timeout=60
context=Main
wrapuptime=15
announce-frequency=30
announce-holdtime=yes
monitor-format=wav49
monitor-join=yes
member = SIP/10061
member = SIP/10062
member = SIP/10063
--8-- queues.conf ---8--
Here is the sip debug:
-- Executing Macro(SIP/10040-681e, call|10027) in new stack
-- Executing Set(SIP/10040-681e, ext=10027) in new stack
-- Executing Dial(SIP/10040-681e, SIP/10027|20|o) in new stack
-- SIP Seeding peer from astdb: '10027' at [EMAIL PROTECTED]:5060
for 3600
12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.20.30.40:5060:
OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK41db2c78;rport
From: Unknown sip:[EMAIL PROTECTED];tag=as3340d5e5
To: sip:[EMAIL PROTECTED]:5060
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 19 Jul 2006 16:02:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 3 lines
Reliably Transmitting (no NAT) to 10.20.30.40:5060:
NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK7776a171;rport
From: Unknown sip:[EMAIL PROTECTED];tag=as0d613efd
To: sip:[EMAIL PROTECTED]:5060
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 94
Messages-Waiting: no
Message-Account: sip:[EMAIL PROTECTED]
Voice-Message: 0/1 (0/0)
---
Scheduling destruction of call
'[EMAIL PROTECTED]' in 15000 ms
We're at 10.20.30.1 port 10656
Video is at 10.20.30.1 port 15268
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 13 lines
Reliably Transmitting (no NAT) to 10.20.30.40:5060:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK723d1311;rport
From: John Doe sip:[EMAIL PROTECTED];tag=as5214182e
To: sip:[EMAIL PROTECTED]:5060
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 19 Jul 2006 16:02:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 9210 9210 IN IP4 10.20.30.1
s=session
c=IN IP4 10.20.30.1
t=0 0
m=audio 10656 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 10027
sip*CLI
-- SIP read from 10.20.30.40:5060:
SIP/2.0 200 OK
To: sip:[EMAIL PROTECTED]:5060;tag=31180d12ce1539b5i0
From: Unknown sip:[EMAIL PROTECTED];tag=as3340d5e5
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.20.30.1:5060;branch=z9hG4bK41db2c78
Server: Linksys/SPA3000-3.1.10(GWd)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY,
[Asterisk-Users] Soundwin S2400 standalone 24FXS/FXO SIP gateways
Hi all, Does anyone on the list have any experience with this piece of hardware? It looks to be another way of bridging * with existing wirings / pstn lines. here is the url: http://www.soundwin.com/s2400.php P.S. it claims to have t38 support too. regards, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF feedthru again...
Doug Crompton wrote: Two other Digium bug reports on this issue. It sure looks like it is an * issue and rather complex??? Any hope for a solution?? http://bugs.digium.com/view.php?id=5970 http://bugs.digium.com/view.php?id=6027 It is an * issue, tested and confirmed. Makes any PSTN side IVR application unaccessible. It is there for the past 2 years. Current rfc2833 implementation in Asterisk requires a total rewrite to fix this. This issue was raised several times, with some answers being an attempt to challenge Sipura's own rfc2833 implementation validity or a promise to fix or submit-your-fix kind of answers. Several possible fixes were posted but never made it to the trunk. If you can leave * out of the path, you wont have any DTMF problems, as soon as * is in the voice path, the dtmf packet sequencing bug manifests itself. You have a choice of disabling all codecs except alaw/ulaw and sticking to the inband dtmf option on both sipura and asterisk sides. HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN outgoing DTMF vs. transfer Problem
Doug Crompton wrote: I am using an SPA-3000 3.1.10d When I have transfer enabled - 'T' in the dial string I cannot reliably send DTMF keys to a bank, voicemail, or other service requiring tones. If I disable (remove transfer option) from the dial string all is fine. I would like to be able to use features but the ability to have reliable DTMF after call completion is more important. Make sure you're using SIP INFO as the DTMF transport. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mpg123 or asterisk
Erick Perez wrote: should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. mpg123 was used back when asterisk didn't have native format support. If you are expecting heavy load, the native format is the way to go. You might decide not to use mp3 format at all, recompressing your MoH files using sox to the formats you gonna use, such as .al, .ul, .gsm, or leave it at .sln to cut the decoding leg only. This way you won't spend expensive CPU power on mp3 decoding and mpg123 processes. also, has anyone succedded in compiling mpg123 in a dual core pentium with centos 4.3 ? No clue here, I've been running Asterisk on FreeBSD systems since 1.0.9. HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using a billing system
exten = _2,1,Answer exten = _2,2,Wait,2 exten = _2,3,DeadAGI, a2billing.php exten = _2,4,Wait,2 exten = _2,5,Hangup I tried it and the call is answered bu Asterisk and never dials the destination. :( Yes that's the correct way to launch A2B script. Are you a2billing.php is in your agi-bin directory? Also, you can see if the script runs without error by executing it from shell(you'll need php cli compiled and installed) and keep pressing enter key to see the script output. Perhaps you have your php binary in the wrong path or a missing php extension. Make sure you have pcntl php extension installed too. HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA 3102 Caller ID in Bellsouth/NA
Julio Arruda wrote: Anyone tried the new PSTN/FXO port in the new SPA 3102 FXO/FXS adapter ? From a quick test (got mine yesterday), seems like it is not recognizing Caller ID from PSTN/FXO port.. It's a known bug in the current firmware. HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?
Andrew Kohlsmith wrote: On Thursday 04 May 2006 11:31, Louis-David Mitterrand wrote: I've got this low-ping 100%-up dsl connection between two asterisk 1.2.7.1 servers. And oftentimes one of them would declare its opposite UNREACHABLE. Same, here, two asterisk 1.2.7.1 boxes connected to the same switch... Over a week I see at least one case of one of the boxes becoming unavailable for the other... simple iax2 reload fixes the problem. Been like this for ages. just my 2 cents, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why a perfectly fine iax2 hostbecomes UNREACHABLE?
On Thu, 4 May 2006 23:29:37 +0200, Louis-David Mitterrand [EMAIL PROTECTED] wrote: On Thu, May 04, 2006 at 10:31:17PM +0500, Vahan Yerkanian wrote: Same, here, two asterisk 1.2.7.1 boxes connected to the same switch... Over a week I see at least one case of one of the boxes becoming unavailable for the other... simple iax2 reload fixes the problem. Is SIP between two asterisk boxes more reliable? Has someone tried it? Yes it is. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk2Billing
Scheda wrote: I'm sure this has been asked a million times. Therefore, I must ask again. Generally speaking, what do you guys think of it. It looks pretty good, but for my uses, I'm not sure that a calling card method is the *best* way to go. But, either way, what is the general concensus? Rock stable, and IMHO the best solution atm. been using for half a year, never had a problem. Kudos to Areski. Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly
This is a known issue with Asterisk's implementation of DTMF detection. There are two bug reports open up on bug tracker. Currently the best combination is to set DTMF TX method on Spa3k to INFO and auto on asterisk side. Works 95%, skips digits if you press buttons on the FXO end too fast. Until the DTMF stuff gets rewritten in Asterisk this gonna be this way, so far 2 years still no 100% dtmf detection, both detection and transmit parts are flawed and dont work 100%, even inband. Affected fxo gateways are (tested by myself): Sipura, Addpac, Planet, Wellgate. HTH, Vahan Dave Fullerton wrote: Greetings, I'm using asterisk to connect our three locations together with a sort of inter-company auto attendant connected like this: PBX (fxs) - Sipura 3k (fxo) - Asterisk -IAX- remote asterisk It works like this: Person picks up their phone and dials a number to get to the auto attendant (I don't have any FXO ports available on our PBX to do it the right way). The attendant answers and asks them the remote extension they want to dial. This setup has worked very well for several months. Last week I upgraded to 1.2.7.1 from 1.2.4 (I think). Since then I've been having trouble with the auto-attendant correctly detecting DTMF (missing digits). Some times it works flawlessly, others I have to try over and over before it is detected correctly. It isn't even consistently dropping the same digit from what I can see on the console. The only thing I've found is that I have a better chance of it working if I wait for the prompt to finish before dialing. I have changed the DTMF method from rfc2833 to info and finally inband with only a little change (inband seems to work the best). Has anyone else run into similar problems or have any more suggestions to try? This is the attendant portion of my extensions.conf: [inter-attendant] exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Set(TIMEOUT(response)=10) exten = s,4,Background(enter-ext-of-person) exten = i,1,Playback(invalid) exten = i,2,Goto(s,4) exten = i,3,Hangup exten = t,1,Playback(goodbye) exten = t,2,Hangup include = tests include = fullertonpbx include = intercompany Thank you for any insight you can provide. Dave Fullerton ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sipura spa2 + asterisk bug ?
Tofik Suleymanov wrote: Thanks all for replying tommorow i'll try to do like you suggested. One more quick question: why Sipuras cant do more than 1 g.729 channel at a time ? Insufficient CPU power to process 2 g729 streams. Is this somehow related to g.729 licensing ? Is there any other SIP adapters which override this drawback ? You're looking for Sipura SPA-2100. It has more powerful CPU. Check http://www.sipura.com/support/spa2100faq/Section_1.html HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sipura spa2 + asterisk bug ?
Tofik Suleymanov wrote: 1. assume 1-st line is in use 2. after dialing 2-nd line from outside i immediately go to the voicemail announcement (also i immediately go to voicemail if i dial from extension to extension both of which are on the same sipura device) Check what response code appears on Asterisk CLI when you dial 2nd line. If your SPA-2k is SPA-2000 or SPA-2002 there might be a codec combination that is still overloading CPU and it's sending back unavailable response. I assume both extensions have separate username/passwords, don't they? Vahan begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 DMTF
Try with dtmfmode=auto and DTMF Tx Method: InBand+INFO, this was the best configuration for me, although still not 100% guarantee. If the dtmf tones are sent very fast without a 1 sec delay, in most of the cases asterisk won't detect half of them. There are a couple of patches for the trunk regarding this issue, but they didn't work for me. HTH, Vahan Chris Mason (Lists) wrote: I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? [snip] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info about F1000G
Tomislav Parčina wrote: Does anybody use UTStarcom F1000G Wi-FI VoIP phone? http://www.utstar.com/Solutions/Handsets/WiFi/ I'm planning to buy one and I need to know did you have any problems with phone. What is the sound quality? How close you need to be to the access point? Please, any information's are useful to me. Bought one as a sample. Build quality is above average. Battery life is very short. Maximum volume of the headpiece is very very low, e.g. you can't hear anything while talking on the street or in a noisy environment like a bar, etc. Menu is not user friendly, sip and network settings are too far from each other. Doesn't support wifi networks that use 802.1x billing systems. Resume: OK phone for home / small office use, but you'll always want to get a better one. Has become a member of Hall of Junk for our HQ museum. ;) HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf
Vladyslav wrote: Just my couple notes on spa3000 and PSTN DTMFs. Such schema: PSTN - SPA3000 - Asterisk Have problems with DTMF detection on incoming calls when call comes from cell phone. Once per 4 times it misdetect some ditigs (whether first digit will be doubled or unrecognized at all). Were tried different (5) cell phones (cell phones providers) Also support from sipura has no clue. Strange, because in my experience the only thing that works 95% is the PSTN - FXO DTMF detection. And cell phones send 1sec DTMF tones in my experience after the call is answered. It's the opposite way that has problems. Just try the following in your setup: 1. Call into your Asterisk system from PSTN. 2. Get connected to an extension with a phone. 3. Press buttons on that phone. 4. What you hear on the PSTN side? What you see on Asterisk console? 5. Press buttons on your PSTN phone. 6. What you hear on your extension phone? What you see on Asterisk console? 7. Change dtmfmode and codec settings in sip.conf for the FXO and extension phone. 8. Goto step 1. You'll see that no codec/dtmfmode setting is 100% accurate. It's easy to blame the phone/adapter manufacturers, but then why the problem vanishes when you have canreinvite=yes for the devices and the Asterisk is out of the path? IMHO this have to be re-inspected and fixed asap. just my 2 cents in, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info about F1000G
wendell hamilton wrote: Ditto on the volume issue, it's EXTREMELY low, but the latest firmware does allow login to web-portal protected billed wifi systems. You don't want web-portal based auth schemes for your metropolitan wifi network, as someone can use your network as a transport with two wifi adapters located on the opposite ends of your city using CIDR ip addresses and eating available bandwidth. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: sipura 841 mass provisioning
C F wrote: On 3/2/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Can you please explain this? In my case, how would I do this: |9,:1xx| |lt;9,:gt;xx| ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf
Actually,
I believe that something is wrong with the way asterisk implements the
whole rfc2833 in rtp.c , moreover, the default value of 100ms in
dtmf_tones[] in do_senddigit() inchannel.c is to short to be detected
for lots of commercially available fxo gateways.
This was reported several times but as of today the issue is there.
I ended up with the following ugly hack for rtp.c channel.c:
--- work/asterisk-1.2.4/rtp.c Wed Mar 1 20:25:03 2006
+++ /root/cvs/rtp.c Tue Feb 21 00:05:55 2006
@@ -1150,7 +1150,7 @@
rtpheader[1] = htonl(rtp-lastdigitts);
rtpheader[2] = htonl(rtp-ssrc);
rtpheader[3] = htonl((digit 24) | (0xa 16) | (0));
- for (x = 0; x 6; x++) {
+ for (x = 0; x 15; x++) {
if (rtp-them.sin_port rtp-them.sin_addr.s_addr) {
res = sendto(rtp-s, (void *) rtpheader, hdrlen
+ 4, 0, (struct sockaddr *) rtp-them, sizeof(rtp-them));
if (res 0)
@@ -1163,12 +1163,12 @@
ntohs(rtp-them.sin_port),
payload, rtp-seqno, rtp-lastdigitts, res - hdrlen);
}
/* Sequence number of last two end packets does not get
incremented */
- if (x 3)
+ if (x 12)
rtp-seqno++;
/* Clear marker bit and set seqno */
rtpheader[0] = htonl((2 30) | (payload 16) |
(rtp-seqno));
/* For the last three packets, set the duration and the
end bit */
- if (x == 2) {
+ if (x == 11) {
#if 0
/* No, this is wrong... Do not increment
lastdigitts, that's not according
to the RFC, as best we can determine */
--- work/asterisk-1.2.4/channel.c Wed Mar 1 20:25:01 2006
+++ /root/cvs/channel.c Tue Feb 21 00:05:50 2006
@@ -2111,22 +2111,22 @@
* it by doing our own generation. (PM2002)
*/
static const char* dtmf_tones[] = {
- !941+1336/100,!0/100, /* 0 */
- !697+1209/100,!0/100, /* 1 */
- !697+1336/100,!0/100, /* 2 */
- !697+1477/100,!0/100, /* 3 */
- !770+1209/100,!0/100, /* 4 */
- !770+1336/100,!0/100, /* 5 */
- !770+1477/100,!0/100, /* 6 */
- !852+1209/100,!0/100, /* 7 */
- !852+1336/100,!0/100, /* 8 */
- !852+1477/100,!0/100, /* 9 */
- !697+1633/100,!0/100, /* A */
- !770+1633/100,!0/100, /* B */
- !852+1633/100,!0/100, /* C */
- !941+1633/100,!0/100, /* D */
- !941+1209/100,!0/100, /* * */
- !941+1477/100,!0/100 }; /* # */
+ !941+1336/250,!0/100, /* 0 */
+ !697+1209/250,!0/100, /* 1 */
+ !697+1336/250,!0/100, /* 2 */
+ !697+1477/250,!0/100, /* 3 */
+ !770+1209/250,!0/100, /* 4 */
+ !770+1336/250,!0/100, /* 5 */
+ !770+1477/250,!0/100, /* 6 */
+ !852+1209/250,!0/100, /* 7 */
+ !852+1336/250,!0/100, /* 8 */
+ !852+1477/250,!0/100, /* 9 */
+ !697+1633/250,!0/100, /* A */
+ !770+1633/250,!0/100, /* B */
+ !852+1633/250,!0/100, /* C */
+ !941+1633/250,!0/100, /* D */
+ !941+1209/250,!0/100, /* * */
+ !941+1477/250,!0/100 }; /* # */
if (digit = '0' digit ='9')
ast_playtones_start(chan, 0,
dtmf_tones[digit-'0'], 0);
else if (digit = 'A' digit = 'D')
Arsen Chaloyan wrote:
Vahan,
see my comments below.
[snip]
Actually SPA-3000 should receive rfc2833 DTMF events
from IP side and put inband DTMFs instead to PSTN
side.
SPA-3000 fails to correctly detect rfc2833 events,
which poorly constructed by asterisk.
So asterisk isn't fully compliant to rfc2833, from the
other hand other VoIP gateways (but not SPA-3000)
manage to detect rfc2833 DTMFs from asterisk.
SPA-3000 serves well in this scenario if source of RTP
stream is another VoIP device or phone (cisco, snom,
aastra...), not asterisk itself.
Here come the solution:
try to exclude asterisk from media path
canreinvite=yes
[111]
;SPA-3000 FXO port
type=friend
username=111
secret=xxx
host=dynamic
canreinvite=yes
dtmfmode=rfc2833
This is the only way I manage to setup
asterisk/SPA-3000 in described scenario.
Hope this will help,
Arsen.
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research Development dept.
adr:;;28, Isahakian ave., PO BOX
Re: [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf
Adding to what I already said, just tested that Asterisk indeed doesn't translate between inband and rfc2833, here is the setup: Analog phone-SPA-3000fxs(g729,rfc2833 friend in sip.conf)-Asterisk- -anotherSPA-3000fxo(g711alaw,inband friend in sip.conf)-PSTN Calling from PSTN, and into the FXS port's extension: a.b.c.d 54321 74acc9957be 00102/0 g729 No Tx: ACK e.f.g.h 123456789 f10b0644-47 00102/00149 alaw No Tx: ACK DTMF sounds get muted within 20-50ms and what is left is high-pitch burst. At the same time, DTMFs from FXS extension sounds same, but generate Mar 1 20:44:56 NOTICE[38801]: rtp.c:565 ast_rtp_read: Unknown RTP codec 101 received Mar 1 20:44:56 NOTICE[38801]: rtp.c:565 ast_rtp_read: Unknown RTP codec 101 received Mar 1 20:44:56 NOTICE[38801]: rtp.c:565 ast_rtp_read: Unknown RTP codec 101 received Mar 1 20:44:56 NOTICE[38801]: rtp.c:565 ast_rtp_read: Unknown RTP codec 101 received Mar 1 20:44:56 NOTICE[38801]: rtp.c:565 ast_rtp_read: Unknown RTP codec 101 received Mar 1 20:44:57 NOTICE[38801]: rtp.c:565 ast_rtp_read: Unknown RTP codec 101 received Mar 1 20:44:57 NOTICE[38801]: rtp.c:565 ast_rtp_read: Unknown RTP codec 101 received Mar 1 20:44:57 NOTICE[38801]: rtp.c:565 ast_rtp_read: Unknown RTP codec 101 received Mar 1 20:44:57 NOTICE[38801]: rtp.c:565 ast_rtp_read: Unknown RTP codec 101 received Mar 1 20:44:57 NOTICE[38801]: rtp.c:565 ast_rtp_read: Unknown RTP codec 101 received Mar 1 20:44:57 NOTICE[38801]: rtp.c:565 ast_rtp_read: Unknown RTP codec 101 received Mar 1 20:44:57 NOTICE[38801]: rtp.c:565 ast_rtp_read: Unknown RTP codec 101 received Resume: while all-g711/inband and all-g729/rfc2833 setups work flawlessly, mixed setups are broken. It's not a Sipura issue too, as adding canreinvite=yes makes the devices communicate correctly. Same thing with other FXO gateways. :S sip.conf: [54321] username=54321 secret=xxx dtmfmode=rfc2833 disallow=all allow=g729 host=dynamic type=friend [spafxo#] username=theusername secret=xxx dtmfmode=inband disallow=all allow=alaw allow=ulaw host=dynamic type=friend begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: sipura 841 mass provisioning
Alan Ferrency wrote: We are mass provisioning Sipura 841's as well. After a normal reboot, our phone retains its previous configuration, so as long as that configuration has not changed, we're fine. However, if we do change the configuration, we have the same issue that you have: it does not pick up the new configuration without a lot of coaxing. Reboot once again and it picks up the new config. Two-step provisioning takes a couple of reboots to insure the device has reconfigured itself. Applies to 2100, 3000, 841 and 941 models. HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SPA-3000 and PSTN dtmf
Greetings, What is the recommended settings for using SPA-3000's FXO port for dialing out to PSTN in regard of the DTMF? The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports registered to the Asterisk box with unique username/passwords. The inbound PSTN DTMF works excellently, e.g. people calling from PSTN into the * box are able to pick IVR items with DTMF reliably. The problem manifests itself when you attempt to place a call via SPA-3000's FXO into PSTN to a different IVR system and try to navigate its menu. Audibly, the other end hears very short DTMF bursts with short silence afterwards, as like SPA-3000 detects a DTMF, mutes it and sends rfc2833 or whatever. Obviously, the burst is short enough to be ignored by the remote IVR system (similar Asterisk/SPA-3000 setup). The relevant settings in the SPA-3000/2100 config for DTMF are set to 'Auto' setting. The relevant lines in sip.conf: ;--- [general] ;irrelevant lines removed dtmfmode=rfc2833 disallow=all allow=alaw allow=ulaw allow=g729 [111] ;SPA-3000 FXO port type=friend username=111 secret=xxx host=dynamic [112] ;SPA-3000 FXS port type=friend username=112 secret=xxx host=dynamic [113] ;SPA-2100 FXS1 port type=friend username=115 secret=xxx host=dynamic [114] ;SPA-3000 FXS2 port type=friend username=114 secret=xxx host=dynamic ;--- Should I instead split the SPA-3000's FXO entry into a type=user and type=peer entries with the first having dtmfmode=rfc2833 and second dtmfmode=inband? Which is the proper way of sending inband dtmf over g711 into the PSTN? Awaiting assistance and thanks for your time, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Welltech USA? and Wellgate Products?
Martin Joseph wrote: Any feedback on this brand and in particular on doing business with WelltechUSA? Don't worry they're OK. I am looking to the Wellgate 3701A which is a 1FXS-1FXO arrangement. I am hoping to replace the near worthless Grandstream HT-488. I'd personally recommend to get a Sipura SPA-3000 instead. You're going to have problems trying to register the FXO port with username/password into Asterisk. Last time I checked, both ports used same CallID, same with the rest of Wellgate products. This company is telling me that I need to wire $ directly into there bank account. Most unusual. We bought all the samples from them via wire transfer too. Samples are collecting dust now though. HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] linksys SPA-941
http://www.sipura.com/support/spa941faq/ has the sample xml for provisioning. Mark Wiater wrote: I asked the [EMAIL PROTECTED] for the documents and the tools that are referenced in the admin guides and was told that I had to become a registered user in the support section of the ww.sipura.com website. They wanted name, title, phone # and type of support I provide for the devices. I think I actually became registered via email with [EMAIL PROTECTED] Got an email the next day with user information for their support site. mark Edwin Lam wrote: does anyone get a hold of the SPA-941 Provisioning Guide? i tried call Sipura's tech support, seems like none of them heard of the term remote provisioning. they kept refering me to their web site which i've check thoroughly, and could not find any documentations on the SPA-941. finally they gave me a phone number to call, which appears to be a fax machine. that's when i gave up on those idiots. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SPA-2100 / 3000 provisioning .xml examples / xml variable list
Hi, I'm looking for a full list of xml provisioning variables of the SPA-2100/3000. Currently the Sipura website has example XMLs only for the SPA-841 [1] and SPA-941 [2]. I'm mostly interested in the CallerID type selector variables and whatever variables control the PSTN-VoIP settings. Sipura Configuration website form field names are numeral only. :( [1] http://www.sipura.com/support/spa841faq/sample-841.xml [2] http://www.sipura.com/support/spa941faq/sample-941.xml Best regards, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : RE : [Asterisk-Users] name that vendor...
welltech... last time i tested their fxo 4 port gateway like year ago all ports were trying to communicate using same Call-ID. [EMAIL PROTECTED] wrote: Sorry, but I don't remember the name of this chinese company. I have meet it once time at a Cebit exhibition at Hannover in Germany few years ago. Francois BERGERET, France. -Message d'origine- De : Jeffery Chen [mailto:[EMAIL PROTECTED] Envoyé : samedi 31 décembre 2005 10:26 À : [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] name that vendor... yes, right ? do your who make this box ? On 31/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hey men, I know this box ! You can see them at : www.ges.fr/voip/ This gateways are exported from Taiwan by Micronet and probably other brand/company. This are made in China and work well (H.323/SIP firmwares). GES is a french distributor and can provide you with a lower price than displayed on their public osCommerce web site for integrators/resellers. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Cory Andrews Envoyé : samedi 31 décembre 2005 04:49 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] name that vendor... Mark - we have never sold this device...just FYI. The only not well known 4FXO device we sell is the ClipComm 4FXO gateway. The rest of the 4FXO devices we offer are from well established companies like Mediatrix and AudioCodes.I deal with the product management side of our business, and from the looks of this device I am not familiar with it at all. Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Mark Phillips wrote: Judicous application of my Staples Easy Button reveals this to be a no name special I Googled it and found the device badged under Ipeya, BossLAN and a whole host of others. Until recently it was on Voipsupply.com too. This is one of the devices that was recently discussed a being a sucky device. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeffery Tel: 1-700-576-1311 FWD: 728150 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stay away from Grandstream!
Stay away from Grandstream and AddPac. These are some of the companies with undereducated software developers that have problems with understanding written english, mainly the SIP RFC documents. I learned this the hard way, wasting half a year with helping them fix problems which shouldn't be there if they have had read/implemented the RFC correctly. Basically, they sell beta quality hardware and then you co-share their final firmware development costs by providing free testing/QA. I blame their sales management for pushing developers to release without proper testing. GXP2000 is much more buggy echo-can wise than the earlier models. For now, I'm back to more expensive equipment. We're not that rich to pay twice. HTH, Vahan Avi Miller wrote: Brian Capouch wrote: They don't perform as well as the expensive Ciscos and Polycoms, but many of us are using them in a variety of circumstances quite happily. I have 4 of them in a small office (GXP2000) running 1.0.12 and they're just fine for our purposes. As Brian said, YMMV. For our 60-person office in Sydney, I'm probably going to use a mix of Polycom/Grandstream and softphones. cYa, Avi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4-port external sip fxo which doesnt suck?
Go with SPA-3000. While it's much more awkward to maintain, they're rock stable and provide the features they advertise for. I'd also add AddPac VoiceFinder series as being not 100% asterisk compatible, expensive and not worth your time (learned this the hard way). It took me 6 months to persuade AddPac that each FXO/FXS has to use unique Call-ID on the same gateway device to work properly with Asterisk and other properly written SIP proxies etc. HTH, Vahan [EMAIL PROTECTED] wrote: I'm looking for a 4-port external sip fxo which doesn't suck. o) Clipcomm CG-410. Poor reviews. o) Mediatrix 1204. Very poor reviews. o) Audiocodes MP104. Poor reviews. o) DLink DVG-3004S. Doesnt seem to exist yet. Is anyone actually using a 4 port external sip fxo which doesn't suck? It almost seems better to buy a pile of SPA-3000 and use them for just SIP FXO. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM2400
Massimo De Nadal wrote: Works perfectly out of the box, almost for my customers :-) The only note is to disable echo training. Could you please elaborate which exact model you're using and what are your opinion about the echo can/training quality? Have you tried spandsp faxing? Thanks in advance, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with 2billing please.
Also try executing /var/lib/asterisk/agi-bin/a2billing.php from the shell, most probably the path to php-cli is wrong or you don't have it installed at all. Jose M. Ramirez wrote: Hi list, all. Please, I need help. Although already I installed a2billing, simply I cannot initiate its execution. Only appears this: -- Executing Answer(SIP/20-456d, ) in new stack -- Executing Wait(SIP/20-456d, 2) in new stack -- Executing DeadAGI(SIP/20-456d, a2billing.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- AGI Script a2billing.php completed, returning 0 -- Executing Wait(SIP/20-456d, 2) in new stack -- Executing Hangup(SIP/20-456d, ) in new stack == Spawn extension (from-internal, 1, 5) exited non-zero on 'SIP/20-456d' -- Executing Macro(SIP/20-456d, hangupcall) in new stack -- Executing ResetCDR(SIP/20-456d, w) in new stack -- Executing NoCDR(SIP/20-456d, ) in new stack -- Executing Wait(SIP/20-456d, 5) in new stack -- Executing Hangup(SIP/20-456d, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/20-456d' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/20-456d' and I cannot happen of there. That lack is it or that I am making incorrect? Regards. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A2billing warnings with new Asterisk 1.2
Rafael R. GV wrote: Hi I have this 3 warnings running a2billling with asterisk new version: a2billing.php|2: -- AGI Script Executing Application: (SetLanguage) Options: (en) Nov 18 12:06:19 WARNING[17440]: pbx.c:5435 pbx_builtin_setlanguage: SetLanguage is deprecated, please use Set(LANGUAGE()=language) instead. a2billing.php|2: UPDATE cc_card SET inuse=inuse+1 WHERE username='7938971' This will change in the coming versions, I hope. Nov 18 12:06:29 WARNING[17440]: file.c:583 ast_readaudio_callback: Failed to write frame -- Playing 'prepaid-enter-dest' (language 'en') a2billing.php|2: RES DTMF : -1 a2billing.php|2: DESTINATION :: -1 a2billing.php|2: APPLY_RULES DESTINATION :: -1 Nov 18 12:06:29 WARNING[17440]: file.c:583 ast_readaudio_callback: Failed to write frame == Spawn extension (default, 19546387993, 2) exited non-zero on 'SIP/46836-08e03d40' The last two warnings appears every time I hungup and I this non-standard stop of the application causes that the card remains in 'in-use' 0 status. rafael These two indicate that one leg of call is non-existent (you hangup). The issue with 'in-use' goes away if you add PCNTL module to your php cli version. HTH, Vahan ___ --Bandwidth and Colocation sponsored by Easynews.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] A2Billing problems. still.
John Fraser wrote: does anybody know what i am doing wrong? help please gzip: stdin: unexpected end of file tar: Read 8572 bytes from Open_A2Billing_version_Raccoon.tar.gz tar: Unexpected EOF in archive tar: Unexpected EOF in archive tar: Error is not recoverable: exiting now asterisk:/usr/src/a2billing# Sounds like a truncated .tar.gz. Make sure the download finishes successfuly. I had no problems downloading the tarball from the website. begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.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] A2billing questions
Rafael R. GV wrote: Hello 1.- I am testing a2billing in a SER-Asterisk implementation but using Mysql versions 4.1.15 and 5.0.15 because I want to integrate its cdr with some tables from Ser database, the a2billing-mysql-schema does not work properly in mysql-5 and in 4.1.15 works well but now I found this issue related to cdr´s: I can see cdr records in 'call' table from mysql console but they don't appear in a2bill web interface, I only can see records of current date ...not sure if its a mysql 4.x incompatibility so now I´ve installed postgres to verify, let me now if someone has same problem. Make sure your cdr table is called 'calls' and not 'call'. This causes a reserved word conflict in mysql 5.0.15, not sure about mysql4. Enable logging in asterisk and raise the debug level from asterisk cli to see what's happening with the INSERT query. 2.- How can I generate invoices in PDF format?, a2billing has the option but it doesnt work for me... do I have to install some aditional software? Probably you're missing the tools for creating the pdf files. Pdflib etc. 3.- I am using caller-id authentication for calls originated in SER but I would like to know if its possible to auto create a card each time a new callerID is found? I´ve already tried with cid_auto_create_card options without success and I think it only works when there is no caller-id present, please confirm that. To my experience, this feature is currently not working. Mr. Areski is aware of the problem and promised to look into it. Regards, Vahan ___ --Bandwidth and Colocation sponsored by Easynews.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] A2billing with Mysql-5.0.15
Rafael R. GV wrote: PS: mysql-schema does not work properly in mysql 5.0.x because only one timestamp with default now() in a table its allowed as you told me and also I´ve found other issue related to auto_increment value: ERROR 1064 (42000) at line 87: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'call ( id BIGINT NOT NULL AUTO_INCREMENT, sessionid CHAR(40) NOT NULL, ' at line 1 This works: CREATE TABLE `calls` ( `id` bigint(20) NOT NULL auto_increment, `sessionid` char(40) NOT NULL, `uniqueid` char(30) NOT NULL, `username` char(40) NOT NULL, `nasipaddress` char(30) default NULL, `starttime` timestamp NOT NULL default CURRENT_TIMESTAMP, `stoptime` timestamp NOT NULL default '-00-00 00:00:00', `sessiontime` int(11) default NULL, `calledstation` char(30) default NULL, `startdelay` int(11) default NULL, `stopdelay` int(11) default NULL, `terminatecause` char(20) default NULL, `usertariff` char(20) default NULL, `calledprovider` char(20) default NULL, `calledcountry` char(30) default NULL, `calledsub` char(20) default NULL, `calledrate` float default NULL, `sessionbill` float default NULL, `destination` char(40) default NULL, `id_tariffgroup` int(11) default NULL, `id_tariffplan` int(11) default NULL, `id_ratecard` int(11) default NULL, `id_trunk` int(11) default NULL, `sipiax` int(11) default '0', `src` char(40) default NULL, PRIMARY KEY (`id`) ) hth, Vahan Yerkanian ___ --Bandwidth and Colocation sponsored by Easynews.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] A2billing with Mysql-5.0.15
Rafael R. GV wrote: thanks Vahan you are right, I have changed 'call t1' for 'calls t1' in balance.php and invoices.php files and then tried to create a new table named 'calls' but mysql 5 has changed syntax for 'TIMESTAMP DEFAULT' and this is the error: - starttime TIMESTAMP DEFAULT 'now()' NOT NULL, this should read starttime TIMESTAMP DEFAULT now() NOT NULL now() is a builtin MySQL function and doesn't need to be enclosed in 's. Also, you can have only one timestamp with default now() in a table... HTH, Vahan begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.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] Format of music for native MoH?
Matt Riddell wrote: Patrick wrote: Hi all, Can anyone please tell me which format music needs to be in for native MoH if my local phones use alaw/ulaw and some gsm g729 connections that come in through the Net. You can have all the codec versions of the moh file. Asterisk shall pick the proper one for the particular channel. ___ --Bandwidth and Colocation sponsored by Easynews.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] A2billing with Mysql-5.0.15
Rafael R. GV wrote:
Hi
I was using a2billing with mysql-4.1.12 and php-5.0.4 very successfully
(thanks to areski for this great project and its invaluable assistance
to solve some issues in my last installation...) now I´ve upgraded mysql
to last release 5.0.15 and, without changes in 'mya2billing' database I
am able to make calls, create and see created cards, etc, but I get
this errors when invoke CDR´s in both admin or user interfase:
*Database error:* Invalid SQL: SELECT t1.starttime, t1.src,
t1.calledstation, t1.destination, t1.sessiontime, t1.username,
t1.terminatecause, t1.sipiax, t1.calledrate, t1.sessionbill FROM call t1
WHERE UNIX_TIMESTAMP(t1.starttime) = UNIX_TIMESTAMP('2005-11-12') ORDER
BY t1.starttime DESC LIMIT 0,25
MySQL 5.0.15 introduces stored functions and procedures that are invoked
by 'call()'. A2Billing uses 'call' for the name of the cdr table. Find
all occurencies of 'call t1' or ' call ' in A2Billing's sql queries and
replace them to 'calls t1' and ' calls '. Don't forget to rename 'call'
table to 'calls'. In short, latest A2Billing doesn't work on mysql
5.0.15 / PHP5 out of box.
On the side note, MySQL doesn't support more than 1 entry with default
value of DEFAULT now() NOT NULL in one table.
regards,
Vahan
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Re: [Asterisk-Users] Problems Installing MPG123 on a 64 Bit System
Jason Becker wrote: Had to do some digging to find out what you were talking about - I guess you are referring to the section Using native Asterisk format_mp3 for Music on Hold* found here: http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf Some of the comments suggest that this solution is far from robust. Would be interested in hearing others experience with this solution for MoH. Not sure what comments you were referring to, but I have native format_mp3 for MoH running on all of my production * servers running FreeBSD 5.4 with Asterisk CVS HEAD since July 2005. No more runaway mpg123 processes, and the mp3 decode quality is great... I've seen up to 70 simultaneous MoH(8khz 16bit mono mp3) calls with about 80% idle on a P4-3Ghz/1Gb ram. begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.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] Problems Installing MPG123 on a 64 Bit System
I'd recommend using native mp3 support that is available in CVS HEAD, as madplayer mp3 decoder gives a lower quality sound (audibly more cranky/noisy). Vahan Jason Becker wrote: Steve Totaro wrote: Anyone know how to get around this? I am stumped. # make mpg123 [ -f mpg123-0.59r.tar.gz ] || fetch http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz [ -d mpg123-0.59r ] || tar xfz mpg123-0.59r.tar.gz make -C mpg123-0.59r linux make[1]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r' make CC=gcc LDFLAGS= \ OBJECTS='decode_i386.o dct64_i386.o decode_i586.o \ audio_oss.o term.o' \ CFLAGS='-DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX \ -DREAD_MMAP -DOSS -DTERM_CONTROL\ -Wall -O2 -m486 \ -fomit-frame-pointer -funroll-all-loops \ -finline-functions -ffast-math' \ mpg123-make make[2]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r' make[3]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r' gcc -DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX -DREAD_MMAP -DOSS -DTERM_CONTROL-Wall -O2 -m486 -fomit-f rame-pointer -funroll-all-loops -finline-functions -ffast-ma th -c -o mpg123.o mpg123.c `-m486' is deprecated. Use `-march=i486' or `-mcpu=i486' instead. cc1: error: CPU you selected does not support x86-64 instruction set make[3]: *** [mpg123.o] Error 1 make[3]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r' make[2]: *** [mpg123-make] Error 2 make[2]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r' make[1]: *** [linux] Error 2 make[1]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r' make: *** [mpg123] Error 2 Use madplayer instead. There are several reasons why Digium the Asterisk community should part ways with mpg123. Regards, ___ --Bandwidth and Colocation sponsored by Easynews.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] WiFi Phones
Andy Hamilton wrote: Anyone have good words to say about any of the WiFi handsets currently available? The UTStarCom F1000 (an 802.11b device) works pretty well. It's about half the $$$ of a Cisco 7920 (which are also pretty nice), but it seems like most of the config is done from the keypad. There is a TFTP option, but it seems that isn't quite perfect. You could check the manual (I programmed the unit without that, except to find that the default password is 88). [snip] The keypad is a touch small, and sometimes I hit the wrong key (and my fingers aren't terribly fat). I also seemed to have a problem transferring calls (using the built in transfer function -- # should still work). Despite many vendors' pages saying that it does 802.1x authentication, it sure looks like WEP is the only available security option. Bought one from VoipSupply too, And yes, it doesn't support 802.1x radius auth (no place to select method, client certificate, etc). I've contacted voipsupply support about this and asked them to remove the 802.1x support listed on the product pages but got a cryptic reply that the phone does support 802.1x MD5.. (md5 is just a method of one of not supported 802.1x auths). Also, the max volume for the headpiece was actually quite low - in noisy environments as on streets you'll have hard time listening to the conversation. Overall, this phone is OK for home and small office use, nothing more. Just my $0.02 in, Vahan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPDATE - 512 Calls w/ Dig Rec: NFS Setup and Benchmarks
Dear Matt, Thanks for your great work and the effort documenting the whole process. I'm sure the whole Asterisk community benefits from this kind of work and it's really something to end up in the wiki. Thumbs up! Best regards, Vahan Matt Roth wrote: List members, My previous post SUCCESS - 512 Simultaneous Calls with Digital Recording documents using a RAM disk to eliminate the I/O bottleneck associated with digitally recording calls via the Monitor application. By recording directly to a RAM disk I was able to maintain good call quality on 512 simultaneous calls. [snip] begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] presence settings and Eyebeam
What is the proper way of adding hints to multiple extensions?
In my case extensions are the same as the sip usernames, while as per
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence
exten = 1234,hint,SIP/1234 works,
exten = _1,hint,SIP/${EXTEN} doesn't. Not sure if I can even use
${EXTEN} here...
Any hints?
Vahan
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research Development dept.
adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
url:http://www.arminco.com/
version:2.1
end:vcard
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Re: [Asterisk-Users] presence settings and Eyebeam
Done. Not sure if picked categories under SIP Mantis correct but here it is: http://bugs.digium.com/view.php?id=5149 VY Olle E. Johansson wrote: File a bug report if it does not work. I think it would be a good idea if it works, even though I usually don't recommend using the extension as the peer name. ;-) /O begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Only single channel recorded with Monitor
Try reinstalling sox - it is responsible for mixing the caller and
callee channels. Also, if IAX2/4506:[EMAIL PROTECTED] is your
real username and password, change them asap, you just made it available
to 1+ people and the archives ;)
Regards,
Vahan
Eric Smith wrote:
We are using the following to record conversations.
exten = _1XXX.,1,SetVar(CALLFILENAME=call_to_${EXTEN:1}_dated_${TIMESTAMP})
exten = _1XXX.,2,Monitor(wav,${CALLFILENAME},m)
exten = _1XXX.,3,Dial(IAX2/4506:[EMAIL PROTECTED]/${EXTEN:1})
exten = _1XXX.,4,Congestion
exten = _1XXX.,104,Congestion
This was working previously to record both sides of the
conversation but now we only have the initiating caller channel
being recorded. Occasionaly the other caller is also recorded
but the speed of the recording is completely wrong causing
distortion and out of sync.
Here fwiw are the logs.
Aug 15 18:31:32 DEBUG[9995]: build_route: Contact hop: sip:[EMAIL
PROTECTED]:5060;line=ikojqrcx
Aug 15 18:31:32 DEBUG[9995]: Device 'SIP/snom' changed to state '2'
Aug 15 18:31:32 VERBOSE[9995]: -- Executing SetVar(SIP/snom-7214,
CALLFILENAME=call_to_00NUMBER_HIDDEN_dated_20050815-183132) in new stack
Aug 15 18:31:32 VERBOSE[9995]: -- Executing Monitor(SIP/snom-7214,
wav|call_to_00NUMBER_HIDDEN_dated_20050815-183132|m) in new stack
Aug 15 18:31:32 VERBOSE[9995]: -- Executing Dial(SIP/snom-7214,
IAX2/4506:[EMAIL PROTECTED]/00NUMBER_HIDDEN) in new stack
Aug 15 18:31:32 VERBOSE[9995]: -- Called 4506:[EMAIL
PROTECTED]/00NUMBER_HIDDEN
Aug 15 18:31:32 DEBUG[9995]: Device 'IAX2/4506/2' changed to state '2'
Aug 15 18:31:32 VERBOSE[9995]: -- Call accepted by 80.127.191.55 (format
G729A)
Aug 15 18:31:32 VERBOSE[9995]: -- Format for call is G729A
Aug 15 18:31:34 VERBOSE[9995]: -- IAX2/4506/2 is ringing
Aug 15 18:31:34 DEBUG[9995]: Ooh, voice format changed to 256
Aug 15 18:31:34 DEBUG[9995]: Ooh, format changed from UNKN to G729A
Aug 15 18:31:45 VERBOSE[9995]: -- IAX2/4506/2 stopped sounds
Aug 15 18:31:45 VERBOSE[9995]: -- IAX2/4506/2 answered SIP/snom-7214
Aug 15 18:31:45 DEBUG[9995]: Stopping retransmission on '[EMAIL PROTECTED]' of
Response 2: Found
Any ideas how to fix this?
Thanks
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research Development dept.
adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
url:http://www.arminco.com/
version:2.1
end:vcard
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[Asterisk-Users] SIP INVITE and caller id / proxy-authorization strange behaviour
Hi all, Today I've stumbled upon a very strange behaviour with an analog fxs/fxo gateway (AddPac AP1002, http://www.addpac.com/english/AP1002.html) connected to a CVS HEAD(from today) Asterisk server. This manifested itself after enabling the CallerID on the pstn lines connected to the FXO ports of the module. Both FXO modules have their own sip username/passwords and are registered to the asterisk box in question, same with the fxs ports on the same device. All of them register from the same ip:port combination. With caller-id disabled, whenever a call comes from pstn to one of the pstn lines, the line is picked up and immediately dialed into an extension on the asterisk box (an ivr menu). Call is authenticated and call flow is ok. For the sake of bandwidth conservation I'm including only the SIP INVITE, I'll post full debug if it's required on request. the sip entry for the FXO port is as follows: [582760] type=friend username=582760 secret=xx host=dynamic qualify=yes the sip invite without the caller-id enabled: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 195.250.76.28:5060;branch=z9hG4bK8d423c2ca428 From: sip:[EMAIL PROTECTED];tag=8d423c2ca4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 28 INVITE Supported: timer, replaces Min-SE: 1800 Date: Mon, 25 Jul 2005 15:04:45 GMT User-Agent: AddPac SIP Gateway Contact: sip:[EMAIL PROTECTED] Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 237 Max-Forwards: 70 v=0 o=582760 1122303885 1122303885 IN IP4 195.250.76.28 s=AddPac Gateway SDP c=IN IP4 195.250.76.28 t=1122303885 0 m=audio 23026 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Now, if I enable the callerid for that port, the caller gets identified, and the following SIP INVITE is sent to the server: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 195.250.76.28:5060;branch=z9hG4bKd8424c26a424 From: sip:[EMAIL PROTECTED];tag=d8424c26a4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 24 INVITE Supported: timer, replaces Min-SE: 1800 Date: Mon, 25 Jul 2005 15:01:44 GMT User-Agent: AddPac SIP Gateway Contact: sip:[EMAIL PROTECTED] Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 240 Max-Forwards: 70 v=0 o=010527911 1122303704 1122303704 IN IP4 195.250.76.28 s=AddPac Gateway SDP c=IN IP4 195.250.76.28 t=1122303704 0 m=audio 23022 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Note the From tag: From: sip:[EMAIL PROTECTED];tag=d8424c26a4 So here it is: it uses the detected caller-id instead of the FXO's username. Still, later on the auth challenge step it resends the invite with the proper auth info: Proxy-Authorization: Digest username=582760, realm=sip.arminco.com, nonce=1886728b, uri=sip:[EMAIL PROTECTED], response=487250bb2f1f17a8b15e9ad727e87a6f, algorithm=MD5 ..asterisk rejects the call with Failed auth on [EMAIL PROTECTED] :( Is there a limitation in Asterisk and it uses the From address as the auth user? This seems buggy.. I'll send the full debugs off-list if someone is interested. regards, Vahan begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] chan_sip.c:939 __sip_xmit warning
Greetings, Since the past week I've started receiving the following warnings on my asterisk servers (FreeBSD / CVS-HEAD). This warning manifests itself with x-lite/x-pro/eyebeam clients as well as sipura devices. All of them have qualify=yes in their settings. Jul 18 22:52:01 WARNING[73576]: chan_sip.c:939 __sip_xmit: sip_xmit of 0x8a3401c (len 483) to 195.x.y.28 returned -1: Address family not supported by protocol family Jul 18 22:52:03 WARNING[73576]: chan_sip.c:939 __sip_xmit: sip_xmit of 0x8a3401c (len 483) to 195.x.y.28 returned -1: Address family not supported by protocol family Jul 18 22:52:05 WARNING[73576]: chan_sip.c:939 __sip_xmit: sip_xmit of 0x8a3401c (len 483) to 195.x.y.28 returned -1: Address family not supported by protocol family Jul 18 22:52:07 WARNING[73576]: chan_sip.c:939 __sip_xmit: sip_xmit of 0x8a3401c (len 483) to 195.x.y.28 returned -1: Address family not supported by protocol family Jul 18 22:52:09 WARNING[73576]: chan_sip.c:939 __sip_xmit: sip_xmit of 0x8a3401c (len 483) to 195.x.y.28 returned -1: Address family not supported by protocol family Jul 18 22:52:11 WARNING[73576]: chan_sip.c:939 __sip_xmit: sip_xmit of 0x8a3401c (len 483) to 195.x.y.28 returned -1: Address family not supported by protocol family sip*CLI show version Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i386 running FreeBSD on 2005-07-17 21:18:39 UTC sip*CLI show version files chan_sip.c File Revision chan_sip.cRevision: 1.781 Any ideas? Best Regards, Vahan ___ 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] CVS HEAD voicemailbox full error
Anyone else has problems with CVS HEAD's from today with voicemail hanging up silently without any debug/error messages when checked? It also keeps insisting that the user's voice mailbox is full and can't store more messages even if I clear/rebuild the /var/spool/asterisk/voicemail stuff. I've tried falling back to voicemail.conf entries from realtime voicemail with the same result. Thanks, Vahan begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] CVS HEAD voicemailbox full error
I just copied an older app_voicemail.so from another * box. :) Mark Edwards wrote: yup. had exactly the same problem as you. been spending the last hour trying to figure out what I did wrong in my config. guess how I fixed it? cd /usr/src/asterisk cvs update make install simple really! ;-) I guess someone posted a bugfix a few mins ago and I just picked it up! ;-) cheers, Mark On 7/14/05, *Vahan Yerkanian* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Anyone else has problems with CVS HEAD's from today with voicemail hanging up silently without any debug/error messages when checked? It also keeps insisting that the user's voice mailbox is full and can't store more messages even if I clear/rebuild the /var/spool/asterisk/voicemail stuff. I've tried falling back to voicemail.conf entries from realtime voicemail with the same result. Thanks, Vahan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: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 -- regards, Mark P. Edwards FWD: 667917 ___ 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 begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] Sipura SPA-841 Volume Oscillation Problem
Just got a reply from sipura support confirming the problem and recommending to use this firmware: http://www.sipura.com/download/temp/phone/spa841-03-01-03-a-vol-fix.zip while they're fixing it and until they release the version 3.1.4 thumbs up for their fast reply. Hugh L. Johnson wrote: I'm having sound quality problems on the remote side with anything higher than 3.1.2(d). 3.1.3(a) oscillates and is just too quiet. the pre-release of 3.1.4(a) is staticy according to multiple folks that I called. ___ 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 begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] Sipura SPA-841 Volume Oscillation Problem
Greetings, I'm experiencing the same problem. It manifests itself mostly in noisy environments - as soon as there is some increase of the ambient noise the volume in the headpiece or the speakerphone decreases immediately, and starts to randomly increase/decrease for some time after the ambient noise gets low. This is 100% repeatable if you start the conversation by using speakerphone. As soon as you switch to the handset, the defect disappears. Now the problem is that 5% of calls via headset have the same problem. I am using the latest firmware for the SPA-841. Javier Ergas wrote: Hi all, The problem is on the volume of the voice sent by the SPA-841. I think the echo cancel algorithm sets a limit to the microphone when detects sounds or noise from the earphone. This problem generates an oscillation on the voice volume sent by the phone and even turns it off completely for very little lapses of time making the communication very uncomfortable. I manage three different implementations with Asterisk and Sipura SPA-841 on different clients and network topologies, and on every one we are experiencing the same situation. Thanks, jergas ___ 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 begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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: Congrats, Europe!
http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136 http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/ begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] SIP/iax devices in Russia
Yes, sipuras work well in Russia. Actually, they're so configurable that I think they'll work everywhere. You'll need to re-configure to make them detect/generate Russian tone standard. snacktime wrote: Will sip/iax devices designed for European use also work in Russia? I'm specifically looking at using the Sipura ata's if anyone can confirm they work. Chris ___ 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 begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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: USB handsets / softphones
Hi all, After googling around and searching both * and xten archives, I was still unable to find a working pair of softphone/usb *handset* that work with both keypad operating the softphones buttons *and* working incoming call ringer on the handset. I'm hoping that, while being OT for * discussion, someone else on this list had luck with finding a pair that works, preferably with xten's xlite/xpro. Any feedback is appreciated. regards, Vahan begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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: USB handsets / softphones
Thanks a tons Kerry, I got spa-1001, 2100, 3000, 841 and other stuff... I totally agree with you, but one of my customers still insists on having a handset connected to his laptop as he doesn't want to have additional devices. Any other hints? Kerry Garrison wrote: Here is just my personal opinion on the whole thing as I spent a good deal of time on this myself. In the end I had MUCH better results, and better sound quality moving to a Sipura SPA-1001 and a $14.99 cordless phone (with $12 rebate at Best Buy). Not only does it sound better, I don't have to walk around carrying my huge laptop. Full review of the SPA-1001 will be on GeekGazette tonight. Kerry http://geekgazette.com begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] about mpg123
Hi, For madplay, install it, then put this into your musiconhold.conf (adjusting the paths, of course): [classes] default = custom:/usr/local/share/asterisk/mohmp3/,/usr/local/bin/madplay -Q -z --fade-in --mono -R 8000 --output=raw:- Subjectively, the quality is a little worse than with mpg123 though. regards, Vahan Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, how could I use rawplayer.c as http://www.voip-info.org/wiki-Asterisk+FreeBSD, or madplayer instead of mpg123? Thank you very much for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCVO0XMakHrsrHP9wRAvw0AJ9cTzDIHrzXe47qiFcCObeVo/IllgCghTRT a3ZY1bgUixvAt/BgutLMFf8= =EuiM -END PGP SIGNATURE- begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] Sipura 2000 x dual g729 channels x other choices?
I confirm too, Sipura devices have flawless g729a codec. Tested personally the Sipura-2100, 3000 and 841 hardphone models - all work with Asterisk 100% straight out of the box, even with chan_sip's not_so_100%_rfc3261 behaviour. I think the sipura-1001 model is the stripped-down 1 fxs port copy of 2100, and as they all share the same firmware core, it should work ok as well. The overheating problem seems gone too, my sipuras are only moderately warm after running continously for several days. Now if Sipura could make an 8 port fxs version... ;-) Michael D Schelin wrote: Don't believe everything you read. There is nothing wrong with the sound quality of the G729 codec on the sipura devices. The 2000 does not support both channels running G729 at the same time. This limitation has be fixed with there new product. I forget the model number. Most G729 sound problems can be traced to busy or poorly designed networks. Too much packet loss. I'm a sip service provider and have seen everything with sip. Supura is the best product on the market today. Hermann Wecke wrote: begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] chan_sip not 100% RFC3665 compliant - re-REGISTERs fail.
Greetings, For the past 2 months I've been struggling with registration problems with asterisk+external FXS/FXO gateways (www.addpac.com) that use RFC3665 re-registration procedure. This problem occured for devices with more than one FXS port with a set non-empty password. Those gateway attempt to re-register after the initial register timeout period expires fully compliant with RFC3665, clause 2.2 (http://www.zvon.org/tmRFC/RFC3665/Output/chapter2.html#sub2), but asterisk fails to authenticate them. The 1st FXS port of the device always registers successfuly (although still uses same RFC3665, clause 2.2 procedure), but the remainder fail miserably. Using an account/username with an empty password for the affected ports fixes the problem - so this is something with www-digest method (?). I've spent 2 weeks debugging this with addpac development team, and the same device authenticates flawlessly with Sonus Proxy Server, SNOM Proxy Server, LongBoard Proxy Server, Nortel Proxy so this seems to be a problem with chan_sip. I'm hesitant to post the long sip debug outputs to the mailing list to conserve the bandwidth. More info and sip debugs are available at http://bugs.digium.com/bug_view_page.php?bug_id=0003726 Is there anyone else with the same problem? regards, Vahan begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] Welltech with Asterisk Registration
Hi, This is a confirmed bug with Welltech 38xx sip fxo and 35xx sip fxs. Their SIP stack is not following SIP RFC and is using same CallID for all ports. Welltech claims to have a new fixed firmware for 35xx SIP FXS device, but they're still working on firmware for 38xx SIP FXO devices (since october 2004 when I reported this). Until then devices are collecting dust on the shelf for me. Anyway, you don't need to register your ports to dial out on fxo - use a secret prefix to prepend the phone numbers you send to the FXO. HTH, regards, Vahan Vice President - Lamsre wrote: Please help me, i can only able to register 1 port of my 6 port fxo (sip) with asterisk, it alway last one register. not all port. how to fix this proble. ___ 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 begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] native MOH with Asterisk 1.0.5 - any news?
Was wondering if there are any news on the native MoH patch for 1.0.3/1.0.5.. or this still works on CVS HEAD only? begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] ANNOUNCEMENT : NEW CallingCard Application forAsterisk
Sergey, You should really revisit MySQL.com :) 4.0.x is way outdated... Regarding the high load etc... how about this copy-pasted excerpt from phpmyadmin? ---8 This MySQL server has been running for 19 days, 6 hours, 8 minutes and 28 seconds. It started up on Jan 07, 2005 at 02:21 PM. Server traffic: These tables show the network traffic statistics of this MySQL server since its startup. Traffic ø per hour Received 2,579 MB 5,714 KB Sent 1,050 MB 2,327 KB Total 3,629 MB 8,040 KB Connections ø per hour % Failed attempts 83 0.18 0.00 % Aborted 1,416,484 3,065.05 6.42 % Total 22,064,865 47,744.87 100.00 % Query statistics: Since its startup, 68,530,509 queries have been sent to the server. ---8 regards, Vahan Sergey Kuznetsov wrote: Robert, It is better to stay with Postgres. If you don't want to loose your business stay away from MySQL. If you are from Toronto ( I suppose you are ), you can check my posts to TLUG (Toronto Linux User Group) regarding MySQL and Postgres. I would say Postgres is a Open Source Oracle. It's very stable, very scalable and it's perfectly works under serious workload. MySQL is dying at the same configuration. I have client of mine who having issue with MySQL. Under some workload ( 10 users inserting at the same time ) it corrupts the index. Even MySQL 4.0.X is still corrupts the indexes under heavy load. I never saw it with Postgres. At the same time Postgres provides you a very flexible SQL language and features, as well as you can make stored procedures on Perl and many-many more. All the Best! Sergey. Robert Augustyn wrote: NICE! I understand that it works against Postgress, any idea what it would take to port it to mysql if anything? robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Areski Sent: Wednesday, January 26, 2005 12:05 PM To: Asterisk-Users Mailing-list Subject: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use of a Cardnumberthe Cardnumber can also be defined as accountcode into sip.conf, iax.conf, etc.. * take care of multiple calls using the same Cardnumber * Caller gets informed about his credit Announce the remaining credit * Caller is requested to enter a destination number * Announce the maximal call time for the given destination number It calculates the remaining duration of the actual call (based on tariffrate tables), informs the caller about this and sets a timeout * Interupt the call if the card balance gets zeroWarn the caller about the call interupt 60 30 seconds before the call gets interupted * It connects the Caller to the destination through the configured trunknote : different trunks can be configured and associated by prefix * After disconnecting the call AGI updates the credit and stores the concerning Call-Detail-Records with CallingPartyNumber, CalledPartyNumber, CallSetupTime, Duration, Charge and the remaining credit FEATURES - WEB INTERFACE: * CARD/CUSTOMERS * List customers * Refill customer * CARD/CUSTOMERS * List customers/cards * Refill customer/card * Create customer/card * Generate customers/cards * BILLING * View money situation * View Payment * Add new Payment * RATECARD * List Tariffplan * Create new Tariffplan * Define Tariffplan * TRUNK * List Trunk * Add Trunk * CALL REPORT - BALANCE Last note : It's distributed under GNU GPL Licence. I hope there will have a big interest for the soft, I am waiting your feedbacks... Regards, /Areski -_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_ Belaïd Arezqui www.areski.net E-mail : areski [EMAIL PROTECTED] gmail (.dot.) 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 ___ 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 begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept.
Re: [Asterisk-Users] Inband DTMF is not supported on codec G.711 u-law. Use RFC2833
http://lists.digium.com/pipermail/asterisk-users/2004-August/059869.html Paul Rodan wrote: Using FireFly, all other codecs but G711 Ulaw is selected. But whenever I place a call, I get: Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Umm, wtf? I thought Inband was ONLY supported on G.711 u-law. begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] Wellgate 3804 Firmware
The login and password are voip/voip Miguel wrote: Where I can find the firmware for the Wellgate 3804 ? The files are: - 2m4sipfxo.103 - 4fxosip.103 I don't have a password to pick up it at the welltech site. Kind regards, Miguel ___ 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 begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] Wellgate 3804 Firmware
That device is complete waste of time and money. I've been contacting their support for the past 3 months and all I could get were promises and late replies. Their SIP firmware is not SIP RFC compliant, as it doesn't follow the Call-ID specification and uses the same one for all ports. Jorge Mendoza wrote: Vahan, Firmware 103 is working for you?, Not for us. Pls advise. Jorge Mendoza Vahan Yerkanian wrote: The login and password are voip/voip Miguel wrote: Where I can find the firmware for the Wellgate 3804 ? The files are: - 2m4sipfxo.103 - 4fxosip.103 I don't have a password to pick up it at the welltech site. Kind regards, Miguel ___ 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 begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ 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] Problem with Grandstream bt100
R A wrote: the problem is this: i plugin the phone but it never wake up. there is something to do Yes, search archives, I've previously given recovery instructions. thanks wert begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Grandstream Firmware 1.0.5.16 Attended Transfer
further digging in the firmware reveals fm.grandstream.com/gs which has some more files including .17 begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] MPG123
As easy as having something like this in your extensions.conf exten = 555,1,Answer exten = 555,2,MP3Player(http://www.yourfavradio.com:port/) exten = 555,3,Hangup Roy Sigurd Karlsbakk wrote: Anyone been able to integrate say ICECast or Shoutcast broadcasts into their MOH... I guess if you used something like xmms (X-Winamp) or something like that you could do it?? I'd like to be able to take a good streaming radio station and make it my MOH.. You can do that with today's MP3Player :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] MPG123
Was showing only the use of mp3player stuff to get a shoutcast stream. if you want this in moh, do the following: 1. create a sep directory inside /var/lib/asterisk or whatever you have configured for that, f.e. /var/lib/asterisk/mohmp3-radio, then 2. touch /var/lib/asterisk/mohmp3-radio/dummy.mp3 3. then add live = mp3:/var/lib/asterisk/mohmp3-radio,http://www.yourfavradio.com:port/ into your /etc/asterisk/musiconhold.conf or wherever it is. 4. change your MoH class to 'live' for this exampls and you're done. works like a charm for me :) Brian West wrote: WRONG you can't do that for hold music. bkw begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] internet bandwidth
As an additional info, G723 is like 1700 bytes per second, GSM is like 4600-4800 bytes per second as viewed by netstat - those numbers are with ip overhead. I've been able to use a dialup as a link to get 2 simultaneous G723-based sip hardphones to * server. kido noagbodji wrote: Hi Hammoud, It all depends on the codec that you are using. Best case scenario is with G723 codec 6.3Kbps per channel * 20, around 126K without the overhead. But you problably won't be able to use this codec unless you are in passthru mode (license is pretty expensive). Using g729 you will be using 8K so a total of 240K+ total bandwidth (passthru OK but you can purchase the license from digium)... Kido - Original Message - *From:* chawki hammoud mailto:[EMAIL PROTECTED] *To:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *Sent:* Thursday, November 18, 2004 7:55 AM *Subject:* [Asterisk-Users] internet bandwidth Hi everybody: How much internet bandwidth and spees is enough to handle twenty simultanous SIP calls. Do you Yahoo!? The all-new My Yahoo! http://my.yahoo.com Get yours free! ___ Asterisk-Users mailing list [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Head, Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Best setup for BudgeTone
Wilson Pickett wrote: I'd like to know what's most reliable configuration for BudgeTone 101 in snip The .16 firmware is beta and it has been found to work poorly for several people, including me. I went back to .5.11 I would try to check that first Exactly, .16 has several bugs like message button not working, but .5.11 has a *nasty* bug with not-reregistering after the timeout period, which leads to phone not ringing on incoming calls - you have to power cycle the phone to get it working. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Best setup for BudgeTone
[EMAIL PROTECTED] wrote: On Sun, 14 Nov 2004, Vahan Yerkanian wrote: Exactly, .16 has several bugs like message button not working, but .5.11 has a *nasty* bug with not-reregistering after the timeout period, which leads to phone not ringing on incoming calls - you have to power cycle the phone to get it working. Urk. I'm about to deploy 70 phones at a client and was intending to use .5.11. Can't say I've noticed this problem in testing. What is the current blessed and recommended version then? Steve .5.11 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Best setup for BudgeTone
Jean-Denis Girard wrote: Well, 1.0.5.16 is the official version on the grandstream site: http://www.grandstream.com/y-downloads.htm. I only installed it last friday, so I'm not sure it is better or worse now. I was using 1.0.5.11 before, and was not aware of the non-reregistering problem, which would explain why the phone would not ring. Now using the static IP the phone no longer need to register, so I may safely go back to 1.0.5.11, right? Unfortunately that's not correct. Try this (with static IP): Set up the phone's re-register delay to a say 5 minutes. Save Reboot. Call once. Phone rings. Wait 6 minutes. Call again. Phone doesn't ring and if you have something other than Dial() for that extension, say voicemail, it activates. My solution was to put a large value for the timeout, and reboot reboot reboot. This is true with 1.0.5.11, not sure about 1.0.5.16, as I rolled back from it as the message button wasn't working, sending only 'INVITE:' instead of the full SIP message to call the voicemail extension. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Grandstream BT100 Message Button
1.0.5.16's Message button is bugged, it only sends 'INVITE:' instead of
the full INVITE message. 1.0.5.11 is the latest fully usable firmware.
Gary White (Network Administrator) wrote:
Can anyone tell me how to get the Granstream Message
Button back workig after upgrading to Firmware 1.0.5.16.
This worked before upgrading. Went back to 1.0.5.11
and it works again.
Phone...
/Voice Mail UserID: *99
Asterisk...
exten = *99,1,VoiceMailMain(s${CALLERIDNUM})
/
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Research Development dept.
adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
url:http://www.arminco.com/
version:2.1
end:vcard
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[Asterisk-Users] adding an artificial delay to *
Greetings, Is there a way to add artificial delay to the rtp stream? Due to regulations in our country, it is required to add 400ms delay to *some* VoIP calls. Is this possible with any module? regards, Vahan begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Wildcard X100P question
One at a time, as X100P is to be connected to a single PSTN phone line with a RJ-11. christophe de coninck wrote: Hey, I knew that info already but the question i ment to ask was: how many calls will I be able to make to the outside from my asterisk server with one X100P card, only one at a time or more ? begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] grandstream 102 flashing
Try connecting the phone with crossover utp cable to a computer, set the ip number of that computer to 67.153.142.69 and 192.168.0.159, setup a tftp server on it, and on the next reboot the phone should get the firmware or you'll be able to sniff more info on what it wants. regards, Vahan dean collins wrote: Hi Sjaak, Sorry I'm not sure what you mean by this? I cant see the dns via the lcd (lcd display non responsive) and unable to log in via web address of 192.168.0.160 either. Thanks for your help, I've just spent 3 hours trying things and about to give up. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sjaak Nabuurs Sent: Thursday, October 21, 2004 4:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] grandstream 102 flashing Check out your DNS settings Wait +60 sec dean collins wrote: Can you (or anyone else out there) tell me how to fix this? Basically what happened was I tried to log into the web interface (ip address allocated by sbs dhcp using mac address) but when I hit login it rebooted and has been doa since then). I checked using MS sbs network monitoring all it seems to be doing is asking for a ARP Rarp request to 67.153.142.69 The other thing is it thinks it is ip address 192.168.1.160 but that isn't even part of my network. Any thoughts on what to do from here? The lcd display is totally non responsive. * From: * [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *BetaTeilchen *Sent:* Wednesday, October 20, 2004 11:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] grandstream 102 flashing This flashing is an indicator for a damaged firmware in your phone. Maybe an interrupted TFTP-Download when powered up or just a wrong firmware. dean collins schrieb: Does anyone know what it means when a grandstream flashes the red key light 5 times repeatedly in cycles? I got a new handset delivered to me today, powered up fine until I tried to access it via the web interface using the password admin and then it rebooted with the lcd never displaying again and the red keys flashing 5 times then a break of 3 seconds then repeat. Cheers, Dean ___ Asterisk-Users mailing list [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Vahan Yerkanian n:Yerkanian;Vahan org:ARMINCO Global Telecommunications;Research Development dept. adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE url:http://www.arminco.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
