[Asterisk-Users] Redirecting to another service/server
Hi guys. Without having a FWD account, can Asterisk redirect calls to FWD? For instance, an extension behind Asterisk dials 99751234, and Asterisk says "that starts with 99. let's strip off the 99 and call 751234 at FWD, IE: sip:[EMAIL PROTECTED]:5060". Is that possible, or would services such as FWD reject the call since the device making the call (Asterisk) hasn't registered? Thanks! -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --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 mp3 which are installed with Asterisk
Zach A wrote:
Hi,
The 3 MP3 files which are installed with asterisk, what is their bit
rate, are they mono and do they have ID3 tags?
Zach A
{192}([EMAIL PROTECTED]:Desktop)# file /var/lib/asterisk/mohmp3/*.mp3
/var/lib/asterisk/mohmp3/fpm-calm-river.mp3: MPEG ADTS, layer III, v1,
128 kBits, 44.1 kHz, JntStereo
/var/lib/asterisk/mohmp3/fpm-sunshine.mp3: MPEG ADTS, layer III, v1,
128 kBits, 44.1 kHz, JntStereo
/var/lib/asterisk/mohmp3/fpm-world-mix.mp3: MPEG ADTS, layer III, v1,
128 kBits, 44.1 kHz, JntStereo
{275}([EMAIL PROTECTED]:Desktop)$ id3tool
/var/lib/asterisk/mohmp3/*.mp3
Filename: /var/lib/asterisk/mohmp3/fpm-calm-river.mp3
No ID3 Tag
Filename: /var/lib/asterisk/mohmp3/fpm-sunshine.mp3
No ID3 Tag
Filename: /var/lib/asterisk/mohmp3/fpm-world-mix.mp3
No ID3 Tag
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
___
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[Asterisk-Users] Info about mp3 which are installed with Asterisk
Hi, The 3 MP3 files which are installed with asterisk, what is their bit rate, are they mono and do they have ID3 tags? Zach A ___ --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] MOH native files
SoX needs that libid3tag, libmad and madplay are installed before it can read mp3 files and convert them into some other format. Zach A -Original Message- From: Chris Stenton [mailto:[EMAIL PROTECTED] Sent: Thursday, March 02, 2006 3:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MOH native files sox -V foo.mp3 -t au -r 8000 -U -b -c 1 foo.ulaw resample -ql Chris ___ --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] Variable
Is there a variable to read to see how many calls are currently open? (related to channel status?) PaulH ___ --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] Polycom 501 power over ethernet
The IP300/301 has the power jack, the IP500/501 the inline cable. PaulH On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote: > Not true. Some do and some don't. Some have a place to plug a separate DC > adapter, and some have the inline power, where the adapter plugs into the > ethernet cable. Not sure which ones are newer, and which are older. > > -Original Message- > From: Michael Welter [mailto:[EMAIL PROTECTED] > Sent: Sun 3/5/2006 6:50 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: > Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet > > > > The IP501 does not have a power jack. You'll need one of the Polycom > cables. > > William M Conlon wrote: > > My recollection of the marketing fluff was that we would just use our > > legacy network (cables) and the devices at both ends would figure out > > whether they were sourcing, sinking, or neither. In the case of the > > 501, it's the special Polycom cable, either with or without provision > > for an AC power adapter, that powers the phone. That's what I meant > by > > saying the '501' itself is not compliant with 802.3af -- it needs a > > separate thingamajig [tech jargon :)]to be powered. > > > > Anyway I had hoped that I could just plug a CAT-5 patch cable from my > > RJ45 wall outlet into the phone. > > > > On Mar 5, 2006, at 5:17 PM, Michael Welter wrote: > > > >> As I understand 802.3af, the phones go through a negotiation with the > >> unit supplying the power. I don't think it's a matter of -48VDC on a > >> particular pair. I remember a schematic from years ago--it had each > >> of the receive pair and the transmit pair going into a transformer > >> winding, and that winding had a center tap for PoE. This is not > >> something that *I* am going to screw with. > >> > >> The IP501 telephone set is the same for both PoE and local power. > >> With the PoE cable, the 802.3af electronics (the negotiator) is a > >> plastic thing in the cable. For the local power, there is a plastic > >> thingie toward the wall end of the cable, and you plug the wall wart > >> into the plastic thingie. here> > >> > >> With local power, there is still only one cable one the desk--the > >> power plugs into the cable towards the wall. Except for a power > >> interruption, this has all the advantages of PoE. > >> > >> > >> > >> William M Conlon wrote: > >>> I saw that Polycom offered a cable (not stocked anywhere), at $40 a > >>> pop for 802.3af connections. That's what made me think the phone > >>> itself is NOT 802.3af compliant. > >>> Presumably, for $40, there's more than a fuse in that special cable. > >>> On Mar 5, 2006, at 4:31 PM, Paul Hales wrote: > For Polycom IP500/501's and IP300/301's you need a special polycom > POE > cable. > > When you buy Polycom phones you can usually specify POE or > powerpack. > > PaulH > > On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: > > When I bought two Polycom 501 SIP phones, I naively thought they > were > > Power-over-Ethernet (IEEE 802.3af) because they were "powered over > > ethernet." Silly me. > > > > Polycom must have some odd voltage or funny way of injecting the > > power, because the POE switch I bought for them (Netgear [EMAIL > PROTECTED]) > > won't power them, though if I use the Polycom-supplied AC adapter > and > > ethernet power injector cable, they work with the switch in either > > its powered or unpowered ports. > > > > Anyhow, I hadn't seen any mention of how people power these > phones, > > as I had planned on centralizing phone power on a UPS to supply my > > Asterisk server and POE switch. Now the question is: > > > > Can the Polycom AC-powered injector be used with a standard > ethernet > > patch cable: > > > > switch :: Polycom injector cable :: RJ45 coupler :: patch > cable :: > > Polycom 501 > > > > which would allow me to power the Polycom AC adapters by my UPS. > Or > > do I need to provide a UPS at each phone and run the ethernet like > > > > switch :: patch cable :: RJ45 coupler :: Polycom injector > cable :: > > Polycom 501 > > > > thanks. > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRI
RE: [Asterisk-Users] Polycom 501 power over ethernet
Not true. Some do and some don't. Some have a place to plug a separate DC adapter, and some have the inline power, where the adapter plugs into the ethernet cable. Not sure which ones are newer, and which are older. -Original Message- From: Michael Welter [mailto:[EMAIL PROTECTED] Sent: Sun 3/5/2006 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet The IP501 does not have a power jack. You'll need one of the Polycom cables. William M Conlon wrote: > My recollection of the marketing fluff was that we would just use our > legacy network (cables) and the devices at both ends would figure out > whether they were sourcing, sinking, or neither. In the case of the > 501, it's the special Polycom cable, either with or without provision > for an AC power adapter, that powers the phone. That's what I meant by > saying the '501' itself is not compliant with 802.3af -- it needs a > separate thingamajig [tech jargon :)]to be powered. > > Anyway I had hoped that I could just plug a CAT-5 patch cable from my > RJ45 wall outlet into the phone. > > On Mar 5, 2006, at 5:17 PM, Michael Welter wrote: > >> As I understand 802.3af, the phones go through a negotiation with the >> unit supplying the power. I don't think it's a matter of -48VDC on a >> particular pair. I remember a schematic from years ago--it had each >> of the receive pair and the transmit pair going into a transformer >> winding, and that winding had a center tap for PoE. This is not >> something that *I* am going to screw with. >> >> The IP501 telephone set is the same for both PoE and local power. >> With the PoE cable, the 802.3af electronics (the negotiator) is a >> plastic thing in the cable. For the local power, there is a plastic >> thingie toward the wall end of the cable, and you plug the wall wart >> into the plastic thingie. >> >> With local power, there is still only one cable one the desk--the >> power plugs into the cable towards the wall. Except for a power >> interruption, this has all the advantages of PoE. >> >> >> >> William M Conlon wrote: >>> I saw that Polycom offered a cable (not stocked anywhere), at $40 a >>> pop for 802.3af connections. That's what made me think the phone >>> itself is NOT 802.3af compliant. >>> Presumably, for $40, there's more than a fuse in that special cable. >>> On Mar 5, 2006, at 4:31 PM, Paul Hales wrote: For Polycom IP500/501's and IP300/301's you need a special polycom POE cable. When you buy Polycom phones you can usually specify POE or powerpack. PaulH On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: > When I bought two Polycom 501 SIP phones, I naively thought they were > Power-over-Ethernet (IEEE 802.3af) because they were "powered over > ethernet." Silly me. > > Polycom must have some odd voltage or funny way of injecting the > power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) > won't power them, though if I use the Polycom-supplied AC adapter and > ethernet power injector cable, they work with the switch in either > its powered or unpowered ports. > > Anyhow, I hadn't seen any mention of how people power these phones, > as I had planned on centralizing phone power on a UPS to supply my > Asterisk server and POE switch. Now the question is: > > Can the Polycom AC-powered injector be used with a standard ethernet > patch cable: > > switch :: Polycom injector cable :: RJ45 coupler :: patch cable :: > Polycom 501 > > which would allow me to power the Polycom AC adapters by my UPS. Or > do I need to provide a UPS at each phone and run the ethernet like > > switch :: patch cable :: RJ45 coupler :: Polycom injector cable :: > Polycom 501 > > thanks. > ___ > --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] static kernel
Hi I run all my Linux boxes without support for kernel modules. I'm in the process of setting up an Asterisk PBX and I want to avoid enabling modules on this box too. Is it possible to compile zaptel drivers statically into the Linux kernel? TIA Paolo ___ --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] Dial() cmd executing Macro - dropped audio
In 1.0.9, if I Dial() with the M option, the specified macro executes just fine, however there is several seconds of silence - no audio transmitted to either caller or callee. After 5 or 6 seconds (in my installation) call audio is transmitted normally. Is this known behavior? Can't seem to find a reference to it. On a different topic, has anyone noticed that google searches of the type "my query about Asterisk" site:lists.digium.com has become increasingly useless? It's like Google is dropping pages from the index or Digium is pulling the pages. Someone's gotta mirror this stuff if this keeps happening. ___ --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] low call volume
lookup info on RX and TX gain on voip-info.org On 3/5/06, billy <[EMAIL PROTECTED]> wrote: > > i have AAH connected to pstn via digium TDM01B > > had been testing it on telewest line (UK cable company) with very little > issues. > now moved to a BT line and had several that i anticipated from infomation on > this list. > the one that has caught me out is low volume from the caller via pstn. > > using sipura spa-941's and have to push the volume up to hear. > is there a setting that can correct this > > thanks > > ___ > --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 > > > -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] incoming calls dropout on PRI over TE110p
Funnily enough upgrading to 1.2.x solved my problems! Well that and optus changing some stuff as well. zaptel-trunk drivers also helped a lot with my echo problems. - Original Message - From: "James Sturges" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Sunday, March 05, 2006 6:52 AM Subject: RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of the site. It is sending CRC errors )to Telsta, drops all calls once a day for 1 second, calls getting stuck, quite unpleasant! I was advised to roll back to 1.0.9 Asterisk, Zaptel and Libpri. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul C Sent: Wednesday, 1 March 2006 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p Paul C wrote: I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP extensions ). SNIP << overlapdial should usually be no in my experience. Okay I've turned that to no with no change. I've just got off the phone to Optus and apparently they had a client in melbourne last week and they fixed the problem by turning crc checking off at the optus end. I don't suppose that was anybody on here ? ___ --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
[Asterisk-Users] Snom 360 Hinting tricks
I was always puzzled by posts to the list about people having problems getting hints to work on a Snom, since I always seem to have no problem making it work. That is, until today when I tried to get a sidecar to work. All I could do was get a monitored extension light to light up continuously, regardless of state. Frustrating! Going back to my working dialplans where I got 1 or 2 lights working fine, I saw the pattern and the difference between working and non-working, and I realized that other people were experiencing the same problem as I was. The trick is the *order* in which you put your hint priorities in your dialplan. My non-working sidecar dialplan had all the hint priorities grouped together: exten => 12345,hint,SIP/12345 exten => 12346,hint,SIP/12346 Which would register the hint, but it wouldn't work on the Snom. The way to make it work, for sure, is to make sure your hint priority is the last priority underneath the *related* priority for the extension. So, this will work: exten => 12345,1,Dial(SIP/12345) exten => 12345,2,Voicemail(u12345) exten => 12345,hint,SIP/12345 exten => 12346,1,Dial(SIP/12346) exten => 12346,2,Voicemail(u12346) exten => 12346,hint,SIP/12346 But this won't: exten => 12345,hint,SIP/12345 exten => 12346,hint,SIP/12346 exten => 12345,1,Dial(SIP/12345) exten => 12345,2,Voicemail(u12345) exten => 12346,1,Dial(SIP/12346) exten => 12346,2,Voicemail(u12346) Also, you will get hooped if you lower-case your "SIP" statements. So SIP/12345 will work but sip/12345 won't. As long as you follow these two tricks above, hint-ing is very straightforward and painless. If you don't, it's really frustrating to get going (as I found out today after a couple of hours of swearing) ___ --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] Polycom 501 power over ethernet
The IP501 does not have a power jack. You'll need one of the Polycom cables. William M Conlon wrote: My recollection of the marketing fluff was that we would just use our legacy network (cables) and the devices at both ends would figure out whether they were sourcing, sinking, or neither. In the case of the 501, it's the special Polycom cable, either with or without provision for an AC power adapter, that powers the phone. That's what I meant by saying the '501' itself is not compliant with 802.3af -- it needs a separate thingamajig [tech jargon :)]to be powered. Anyway I had hoped that I could just plug a CAT-5 patch cable from my RJ45 wall outlet into the phone. On Mar 5, 2006, at 5:17 PM, Michael Welter wrote: As I understand 802.3af, the phones go through a negotiation with the unit supplying the power. I don't think it's a matter of -48VDC on a particular pair. I remember a schematic from years ago--it had each of the receive pair and the transmit pair going into a transformer winding, and that winding had a center tap for PoE. This is not something that *I* am going to screw with. The IP501 telephone set is the same for both PoE and local power. With the PoE cable, the 802.3af electronics (the negotiator) is a plastic thing in the cable. For the local power, there is a plastic thingie toward the wall end of the cable, and you plug the wall wart into the plastic thingie. With local power, there is still only one cable one the desk--the power plugs into the cable towards the wall. Except for a power interruption, this has all the advantages of PoE. William M Conlon wrote: I saw that Polycom offered a cable (not stocked anywhere), at $40 a pop for 802.3af connections. That's what made me think the phone itself is NOT 802.3af compliant. Presumably, for $40, there's more than a fuse in that special cable. On Mar 5, 2006, at 4:31 PM, Paul Hales wrote: For Polycom IP500/501's and IP300/301's you need a special polycom POE cable. When you buy Polycom phones you can usually specify POE or powerpack. PaulH On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power injector cable, they work with the switch in either its powered or unpowered ports. Anyhow, I hadn't seen any mention of how people power these phones, as I had planned on centralizing phone power on a UPS to supply my Asterisk server and POE switch. Now the question is: Can the Polycom AC-powered injector be used with a standard ethernet patch cable: switch :: Polycom injector cable :: RJ45 coupler :: patch cable :: Polycom 501 which would allow me to power the Polycom AC adapters by my UPS. Or do I need to provide a UPS at each phone and run the ethernet like switch :: patch cable :: RJ45 coupler :: Polycom injector cable :: Polycom 501 thanks. ___ --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 Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.com ___ --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 --Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --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 Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.com ___ --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 -- Michael Welter Telecom Matt
Re: [Asterisk-Users] Polycom 501 power over ethernet
My recollection of the marketing fluff was that we would just use our legacy network (cables) and the devices at both ends would figure out whether they were sourcing, sinking, or neither. In the case of the 501, it's the special Polycom cable, either with or without provision for an AC power adapter, that powers the phone. That's what I meant by saying the '501' itself is not compliant with 802.3af -- it needs a separate thingamajig [tech jargon :)]to be powered. Anyway I had hoped that I could just plug a CAT-5 patch cable from my RJ45 wall outlet into the phone. On Mar 5, 2006, at 5:17 PM, Michael Welter wrote: As I understand 802.3af, the phones go through a negotiation with the unit supplying the power. I don't think it's a matter of -48VDC on a particular pair. I remember a schematic from years ago--it had each of the receive pair and the transmit pair going into a transformer winding, and that winding had a center tap for PoE. This is not something that *I* am going to screw with. The IP501 telephone set is the same for both PoE and local power. With the PoE cable, the 802.3af electronics (the negotiator) is a plastic thing in the cable. For the local power, there is a plastic thingie toward the wall end of the cable, and you plug the wall wart into the plastic thingie. jargon here> With local power, there is still only one cable one the desk--the power plugs into the cable towards the wall. Except for a power interruption, this has all the advantages of PoE. William M Conlon wrote: I saw that Polycom offered a cable (not stocked anywhere), at $40 a pop for 802.3af connections. That's what made me think the phone itself is NOT 802.3af compliant. Presumably, for $40, there's more than a fuse in that special cable. On Mar 5, 2006, at 4:31 PM, Paul Hales wrote: For Polycom IP500/501's and IP300/301's you need a special polycom POE cable. When you buy Polycom phones you can usually specify POE or powerpack. PaulH On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power injector cable, they work with the switch in either its powered or unpowered ports. Anyhow, I hadn't seen any mention of how people power these phones, as I had planned on centralizing phone power on a UPS to supply my Asterisk server and POE switch. Now the question is: Can the Polycom AC-powered injector be used with a standard ethernet patch cable: switch :: Polycom injector cable :: RJ45 coupler :: patch cable :: Polycom 501 which would allow me to power the Polycom AC adapters by my UPS. Or do I need to provide a UPS at each phone and run the ethernet like switch :: patch cable :: RJ45 coupler :: Polycom injector cable :: Polycom 501 thanks. ___ --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 Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.com ___ --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 -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --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 Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.com ___ --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] Polycom 501 power over ethernet
As I understand 802.3af, the phones go through a negotiation with the unit supplying the power. I don't think it's a matter of -48VDC on a particular pair. I remember a schematic from years ago--it had each of the receive pair and the transmit pair going into a transformer winding, and that winding had a center tap for PoE. This is not something that *I* am going to screw with. The IP501 telephone set is the same for both PoE and local power. With the PoE cable, the 802.3af electronics (the negotiator) is a plastic thing in the cable. For the local power, there is a plastic thingie toward the wall end of the cable, and you plug the wall wart into the plastic thingie. With local power, there is still only one cable one the desk--the power plugs into the cable towards the wall. Except for a power interruption, this has all the advantages of PoE. William M Conlon wrote: I saw that Polycom offered a cable (not stocked anywhere), at $40 a pop for 802.3af connections. That's what made me think the phone itself is NOT 802.3af compliant. Presumably, for $40, there's more than a fuse in that special cable. On Mar 5, 2006, at 4:31 PM, Paul Hales wrote: For Polycom IP500/501's and IP300/301's you need a special polycom POE cable. When you buy Polycom phones you can usually specify POE or powerpack. PaulH On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power injector cable, they work with the switch in either its powered or unpowered ports. Anyhow, I hadn't seen any mention of how people power these phones, as I had planned on centralizing phone power on a UPS to supply my Asterisk server and POE switch. Now the question is: Can the Polycom AC-powered injector be used with a standard ethernet patch cable: switch :: Polycom injector cable :: RJ45 coupler :: patch cable :: Polycom 501 which would allow me to power the Polycom AC adapters by my UPS. Or do I need to provide a UPS at each phone and run the ethernet like switch :: patch cable :: RJ45 coupler :: Polycom injector cable :: Polycom 501 thanks. ___ --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 Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.com ___ --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 -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --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] Polycom 501 power over ethernet
I saw that Polycom offered a cable (not stocked anywhere), at $40 a pop for 802.3af connections. That's what made me think the phone itself is NOT 802.3af compliant. Presumably, for $40, there's more than a fuse in that special cable. On Mar 5, 2006, at 4:31 PM, Paul Hales wrote: For Polycom IP500/501's and IP300/301's you need a special polycom POE cable. When you buy Polycom phones you can usually specify POE or powerpack. PaulH On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power injector cable, they work with the switch in either its powered or unpowered ports. Anyhow, I hadn't seen any mention of how people power these phones, as I had planned on centralizing phone power on a UPS to supply my Asterisk server and POE switch. Now the question is: Can the Polycom AC-powered injector be used with a standard ethernet patch cable: switch :: Polycom injector cable :: RJ45 coupler :: patch cable :: Polycom 501 which would allow me to power the Polycom AC adapters by my UPS. Or do I need to provide a UPS at each phone and run the ethernet like switch :: patch cable :: RJ45 coupler :: Polycom injector cable :: Polycom 501 thanks. ___ --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 Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.com ___ --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] Polycom 501 power over ethernet
I guess the way you want to do it should work, (over a long run you might run into trouble, but only trial and error will confirm this). However keep in mind that the polycom cables come keyed on one end of the RJ45, so that you don't by mistake put the powered end into the switch. What that means is that you will have to cut that bulging tip off, othewise you shouldn't have a problem. On 3/5/06, William M Conlon <[EMAIL PROTECTED]> wrote: > When I bought two Polycom 501 SIP phones, I naively thought they were > Power-over-Ethernet (IEEE 802.3af) because they were "powered over > ethernet." Silly me. > > Polycom must have some odd voltage or funny way of injecting the > power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) > won't power them, though if I use the Polycom-supplied AC adapter and > ethernet power injector cable, they work with the switch in either > its powered or unpowered ports. > > Anyhow, I hadn't seen any mention of how people power these phones, > as I had planned on centralizing phone power on a UPS to supply my > Asterisk server and POE switch. Now the question is: > > Can the Polycom AC-powered injector be used with a standard ethernet > patch cable: > > switch :: Polycom injector cable :: RJ45 coupler :: patch cable :: > Polycom 501 > > which would allow me to power the Polycom AC adapters by my UPS. Or > do I need to provide a UPS at each phone and run the ethernet like > > switch :: patch cable :: RJ45 coupler :: Polycom injector cable :: > Polycom 501 > > thanks. > ___ > --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] Polycom 501 power over ethernet
For Polycom IP500/501's and IP300/301's you need a special polycom POE cable. When you buy Polycom phones you can usually specify POE or powerpack. PaulH On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: > When I bought two Polycom 501 SIP phones, I naively thought they were > Power-over-Ethernet (IEEE 802.3af) because they were "powered over > ethernet." Silly me. > > Polycom must have some odd voltage or funny way of injecting the > power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) > won't power them, though if I use the Polycom-supplied AC adapter and > ethernet power injector cable, they work with the switch in either > its powered or unpowered ports. > > Anyhow, I hadn't seen any mention of how people power these phones, > as I had planned on centralizing phone power on a UPS to supply my > Asterisk server and POE switch. Now the question is: > > Can the Polycom AC-powered injector be used with a standard ethernet > patch cable: > > switch :: Polycom injector cable :: RJ45 coupler :: patch cable :: > Polycom 501 > > which would allow me to power the Polycom AC adapters by my UPS. Or > do I need to provide a UPS at each phone and run the ethernet like > > switch :: patch cable :: RJ45 coupler :: Polycom injector cable :: > Polycom 501 > > thanks. > ___ > --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] Polycom 501 power over ethernet
When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power injector cable, they work with the switch in either its powered or unpowered ports. Anyhow, I hadn't seen any mention of how people power these phones, as I had planned on centralizing phone power on a UPS to supply my Asterisk server and POE switch. Now the question is: Can the Polycom AC-powered injector be used with a standard ethernet patch cable: switch :: Polycom injector cable :: RJ45 coupler :: patch cable :: Polycom 501 which would allow me to power the Polycom AC adapters by my UPS. Or do I need to provide a UPS at each phone and run the ethernet like switch :: patch cable :: RJ45 coupler :: Polycom injector cable :: Polycom 501 thanks. ___ --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] Preferred editor(s) dialplan coding?
Comments inline: >a vim user myself. I don't use most of what you descvribe below, >however: > >> 1. Syntax Highlighting, and ease of updating that highlighting > >Update asterisk.vim Good idea, primary issue being I'd have to learn vim, but it's looking like a LOT of people agree with the concept of it being a phenomenal editor... Guess I missed the boat ;) >> 2. Auto-updating lists (like sidebars) with: (this is a total >> WISH list) >> Variables >> Contexts >> a Command list? > >Not sure how to implement this in vim > >Maybe through some tweak of ctags? This is commonly seen in Winblows editors, although possibly in X. I've seen it in my Eclipse-based Trustudio editor. Thanks for your input! I'm gonna dig into vim I think :) Sherwood ___ --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] ZapATA channels up, but calls cannot be made
I have a issue with two Zap clone cards where they used to work.I am using [EMAIL PROTECTED] 2.5 which includes Asterisk 1.24 and Zaptel drivers 1.2.4. The system is a new Intel Celerion machine. I used to have the same cards running in a Intel PIII system. in this system, they worked. In this older system, I was able to call into the machine and call out from it. Now that I have upgraded the entire box, I cannot figure out why the cards are not working or, more correctly, I cannot place or receive calls on them.First, a bit of background. I have been running this new machine for a few weeks now connecting to the PSTN through an IAX provider. I have some 10 phones connected to Asterisk using several DLink DVG-1120M VOIP "routers". Voicemail, paging, etc. all works. I can call into the IAX trunk and call any phone. I have two POTS lines which I intend to keep - the IAX provider charges by the minutes, thus, I would like to use this connection only as a backup when the POTS lines are in use. The Asterisk system is replacing a Nortel Venture setup. So, I am at the final stage of implementation - get the Zap cards working, sell the Venture phone system.Now, I have installed the drivers, setup the /etc/zapata.conf and /etc/asterisk/zapata-auto.conf, etc. I have configured Asterisk with two trunks Zap/1 and Zap/2. For testing, I have setup asterisk to sent calls with the dialing prefix of 9 to the Zap trunks. When I attempt to place a call, I receive the following message:Mar 5 18:20:29 NOTICE[13927] app_dial.c: Unable to create channel of type 'ZAP' (cause 0 - Unknown)...on both channels. Neither channel even tried to pickup the line. Anyone have any thoughts of where to look?Here are some of details that may be useful:Cards:02:0b.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface02:0c.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface/etc/zaptel.conf:# Span 1: WCFXO/0 "Generic Clone Board 1" fxsks=1# Span 2: WCFXO/1 "Generic Clone Board 2" RED fxsks=2# Global dataloadzone = usdefaultzone = us/etc/asterisk/zatata-auto.conf (as generated by genzapataconf):;callerid=asreceived; Span 1: WCFXO/0 "Generic Clone Board 1" signalling=fxs_ks; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 1context=from-pstngroup=0channel => 1; Span 2: WCFXO/1 "Generic Clone Board 2" RED signalling=fxs_ks; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 2context=from-pstngroup=0channel => 2zap show status:asterisk2*CLI> zap show status Description Alarms IRQ bpviol CRC4 Generic Clone Board 1 RED 0 0 0 Generic Clone Board 2 RED 0 0 0 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 (FYI: The Alarms go to OK when the PSTN lines are connected - they were disconnected when I ran this command)zap show channels:sterisk2*CLI> zap show chchannel channels asterisk2*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo from-pstn en /extension_additional.conf says:OUT_2 = ZAP/2OUT_1 = ZAP/1(changing these to Zap/1 or zap/1 makes no difference)Anyone have any thoughts on what I am missing? This must be something simple.Mark Buckaway [EMAIL PROTECTED]http://homepage.mac.com/mark.buckawayBlog: http://homepage.mac.com/mark.buckaway/blog---Positive thinking – should never be a substitute for action ___ --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: Asterisk Question
Find perl code attached:
while ($count <= $BACK)
{
print STDERR "$count\n";
@item = pop(@text);
print STDERR "@item\n";
$count++;
}
regards,
PaulH
On Sat, 2006-03-04 at 07:54 -0800, Michael Collins wrote:
> > I actually got it all working - but it's great to see where we did the
> > same
> > thing, and where we differ.
> >
> > I ended up using the 'pop' perl command - inside a loop to go back one
> > item
> > at a time through my list
> >
> > PaulH
>
> Nice work! Perl = TMTOWTDI = There's More Than One Way To Do It
> -MC
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Re: [Asterisk-Users] Dialplan - strip IDD prefix and insert another
Thank you. works like a charm. I'm using [EMAIL PROTECTED] so I had to massage
it into AMP's structure.
Your example is actually the reverse of what I needed to do, but that's not
the issue.
AMP uses a macro to dial (syntax almost exactly the same).
I feel this should be documented somewhere (been googling all day) - so much
appreciated.
- Original Message -
From: "Ira" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, March 05, 2006 21:01
Subject: Re: [Asterisk-Users] Dialplan - strip IDD prefix and insert another
At 07:57 AM 03/05/2006, you wrote:
How can I "strip" the 00 and insert 011 in one entry in the dialplan. I'm
stripping the 00 and passing the rest of the numbers for numbers dialled
as 001X. (as in: 00|1XX.) but in case of numbers out of the US, how would
I insert the 011 ?
exten => _011X. , 1, dial(sip/1/00${EXTEN:3})
Or something similar to that. Match to the 011, delete it, {EXTEN:3}, and
then add the 00 before dialing.
Ira
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[Asterisk-Users] # (send immediately) and dialplan broken on PAP2?
We have a bunch of PAP2s, and using the # to send immediately does not work as described in the manual. The PAP still waits for the "Interdigit_Short_Timer" to expire before sending the dial string. In addition, the dialplan does not cause the string to be sent immediately as it should. Here's the dialplan I'm using: (*x.|xxx|[3469]11|[2-9]xxS0|1[2-9]xx[2-9]xxS0|.) We've seen this behavior with both firmware 3.1.3 and 3.1.9. Has anyone else experienced this? Thanks, Barton ___ --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 in Egypt
Hi, I'm going to send a Sipura SPA-3000 to one of my friends in Egypt. Does anybody has experienced any difficulties configuring the SPA-3000 to meet the Egyptian PSTN network norms. Appreciate your help. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Signate Intro to * - London Training March 21-23
We still have a seat open in the London Introduction to Asterisk class. TKS Paul Paul Mahler [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of McQuiggan, Mark xt46480 Sent: Sunday, March 05, 2006 12:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problem with libpri? While testing a problem with "spontaeously" and "occasionally" rebooting asterisk, I came upon this problem: Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1210770512 (LWP 11346)] 0x002e3fe1 in pri_release_timeout (data="" at q931.c:2589 2589 q931.c: No such file or directory. in q931.c q931.c is in libpri, function pri_release_timeout, and line 2589 reads: if (pri->debug & PRI_DEBUG_Q931_STATE) pri_message(pri, "Timed out looking for release complete\n"); PRI Debug was not on in the asterisk console. Any ideas? My asterisk restarts about twice a day, and drops any current calls in the process. Regards, Mark McQuiggan This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system. ___ --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] Preferred editor(s) dialplan coding?
[EMAIL PROTECTED] wrote: On Sun, 5 Mar 2006, Michiel van Baak wrote: On 21:22, Sat 04 Mar 06, C F wrote: vi here vim :) Combined with the syntax file for asterisk. http://www.bemroses.net/images/curves.jpg -Dan Rotfl!!! Looks like whoever drew the emacs curve couldn't program himself out of a loop in emacs-lisp ;) -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Problem with libpri?
While testing a problem with "spontaeously" and "occasionally" rebooting asterisk, I came upon this problem: Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1210770512 (LWP 11346)] 0x002e3fe1 in pri_release_timeout (data="" at q931.c:2589 2589 q931.c: No such file or directory. in q931.c q931.c is in libpri, function pri_release_timeout, and line 2589 reads: if (pri->debug & PRI_DEBUG_Q931_STATE) pri_message(pri, "Timed out looking for release complete\n"); PRI Debug was not on in the asterisk console. Any ideas? My asterisk restarts about twice a day, and drops any current calls in the process. Regards, Mark McQuiggan This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system. ___ --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] about operator
Andrea, Thinking back to your question, Andrea, I'm really wondering whether or not software solutions like FOP could or would ever scale to serve, for example, a 200 seats, full time receptionist. Obviously, software is flexible and with a suitable keyboard and a smart software-hardware integration, it seems technically achievable to provide in the long run, a descent receptionist solution. But I think drivers to make this happen are simply missing. Some companies sell headphones, others sell monitors, other sell telephony solutions but I'm not aware of any company getting a large share of its revenues from receptionists. So, my opinion is that no one is really investing time and money to design Asterisk or multivendor receptionist tools or products. Regards - Original Message - From: <[EMAIL PROTECTED]> To: Sent: Thursday, March 02, 2006 10:22 AM Subject: Re: [Asterisk-Users] about operator I am sorry, but I don't understand the answer. At least in Italy Human resources department doesn't undertand a bit about Hardware supported by asterisk. We are moving a medium factory from a traditional pbx to an asterisk solution. The Human operator now has a kind of hardware. I would like to know which new kind of hardware he'll have. The software solution provided by FOP, as said by Bartosz Piec, is not bad, but I think that a kind of hw device like the ones that are usually present at the operator seat could be better. I will check the solution suggested by "Olivier Krief" <[EMAIL PROTECTED]> (thank you very much !) Andrea "C F" <[EMAIL PROTECTED] m> To Sent by: "Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion" [EMAIL PROTECTED] m.com cc Subject 01/03/2006 14.10 Re: [Asterisk-Users] about operator Please respond to Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED] ists.digium.com> I think this question can only be answered by Human Resources department. On 3/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: I would like to know which kind of solutions are available, both software and hardware, for human operator in an asterisk environment. The operator should be able to provide the basic standard operation, like to transfer calls and to see if the extensions are busy or not and so on. Thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --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
Re: [Asterisk-Users] How to route incoming calls to different contexts?
what about this?
[incoming]
exten => DID1,1,Goto(incoming1,${EXTEN},1)
exten => DID2,1,Goto(incoming2,${EXTEN},1)
Julian.
On 3/5/06, Tele Cost Price Reducer <[EMAIL PROTECTED]> wrote:
>
> hi Zach,
> i would use GOTOIF to forward the DID from within the [incoming] context to
> the other context. i would try :
> exten => gotoif($[did]=DID1,goto did1|s|1,)
> exten => gotoif($[did]=DID2,goto did2|s|1,)
>
>
>
>
>
> On 3/4/06, Zach A <[EMAIL PROTECTED]> wrote:
> > Both DIDs are SIP and from the same provider. Format of registration is
> > like this:
> >
> > sip.conf
> >
> > [general]
> > bindaddr=xxx.xxx.xxx.xxx
> > port=5060
> > context=incoming
> > disallow=all
> > allow=g726
> > allow=ulaw
> > allow=alaw
> > allow=gsm
> > dtmfmode=rfc2833
> > canreinvite=no ; required for incoming calls to ring extensions
> > insecure=invite ; outgoing call not working without this
> > tos=0x18
> > nat=yes
> >
> > register=DID1:[EMAIL PROTECTED]
> > register= DID2:[EMAIL PROTECTED]
> >
> > [DID1]
> > username=DID1
> > type=peer
> > secret=1234
> > host=xxx.xxx.xxx.xxx
> > fromuser=DID1
> >
> > [DID2]
> > username=DID2
> > type=peer
> > secret=1234
> > host=xxx.xxx.xxx.xxx
> > fromuser=DID2
> >
> > Now both DIDs are sent to context [incoming] which is the default
> > context for SIP. If I add context=incoming2 under any DID section, it
> > doesn't go to that context and still go to the default context. How can
> > I direct DID2 to [incoming2] context?
> >
> > Zach A
> >
> >
> > -Original Message-
> > From: Joseph Tanner [mailto:[EMAIL PROTECTED]
> > Sent: Friday, March 03, 2006 7:18 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] How to route incoming calls to
> > differentcontexts?
> >
> > First, tell us if it's sip, iax, or zap. Then tell us what provider
> > (most will use the same general config, but some like ipkall are
> > special and a bit tricky).
> >
> > joseph Tanner
> >
> > On 3/3/06, Zach A <[EMAIL PROTECTED]> wrote:
> > >
> > >
> > >
> > > Hi everybody,
> > >
> > >
> > >
> > > It should be a simple thing to do but I don't know how to do it. Now I
> > have
> > > 2 DIDs and I want one of them go to [context1] and other one to go to
> > > [context2]. How can I achieve this.
> > >
> > >
> > >
> > > Thanks,
> > >
> > >
> > >
> > > Zach A
> > > ___
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> > >
> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >
> > >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> > >
> > ___
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> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > ___
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> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> ___
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>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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[Asterisk-Users] RE: Asterisk-Users Digest, Vol 20, Issue 31
Being a sixTel customer I can tell you how sixTel bills. They charge $X.XX per month for a DID, they also charge per minute inbound (a certain rate) and they charge outbound at another rate. -Original Message- Date: Sun, 5 Mar 2006 11:28:16 -0500 From: "VIC IP Communications" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] re: Sixtel Services To: Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" Hi, Companies like DIDx and Sixtel, when they state DIDs at $XX.XX per month and $XX.XX per minute/monthly, do these companies provide inbound and outbound routing of calls, or are these rates strictly for inbound Call routing of DIDs? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] low call volume
i have AAH connected to pstn via digium TDM01B had been testing it on telewest line (UK cable company) with very little issues.now moved to a BT line and had several that i anticipated from infomation on this list.the one that has caught me out is low volume from the caller via pstn. using sipura spa-941's and have to push the volume up to hear.is there a setting that can correct this thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialplan - strip IDD prefix and insert another
At 07:57 AM 03/05/2006, you wrote:
How can I "strip" the 00 and insert 011 in one entry in the
dialplan. I'm stripping the 00 and passing the rest of the numbers
for numbers dialled as 001X. (as in: 00|1XX.) but in case of
numbers out of the US, how would I insert the 011 ?
exten => _011X. , 1, dial(sip/1/00${EXTEN:3})
Or something similar to that. Match to the 011, delete it, {EXTEN:3},
and then add the 00 before dialing.
Ira
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RE: [Asterisk-Users] re: Sixtel Services
Inbound should be free as far as I am concerned unless you have a toll free number. Thanks, Steve Totaro _ From: VIC IP Communications [mailto:[EMAIL PROTECTED] Sent: Sunday, March 05, 2006 11:28 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] re: Sixtel Services Hi, Companies like DIDx and Sixtel, when they state DIDs at $XX.XX per month and $XX.XX per minute/monthly, do these companies provide inbound and outbound routing of calls, or are these rates strictly for inbound Call routing of DIDs? Thanks. ___ --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] to configure asterisk to work with the nathelper module of openser
Hi all I'm a newbie in asterisk.I ant to know how i ca configure asterisk to work with the nathelper module of openser to fix the nat problem! Thanks in advance! bets regards Serge Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez la version beta.___ --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] Problem compiling ztdummy on centos 4, 2.6 kernel
Did that too, same errors Take care, Sina -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Joseph Sent: Sunday, March 05, 2006 7:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel make linux26 make install worked for me thanks --- Dovid Bender <[EMAIL PROTECTED]> wrote: > did you uncommnet # from before ztdummy ? > > --- Sina Bahram <[EMAIL PROTECTED]> wrote: > > > Hi all, > > > > I hope everyone is doing well. I just joined the list, and I've > > really enjoyed all I have read about asterisk so far. > > Unfortunately, I'm having a > > bit of trouble implementing this thing :). > > > > By the way ... I did my best to search the forums, and also to use > > google extensively, and while I have found pages with people with > > the same problem, ... The fix suggested on those sites, didn't work > > for me. > > > > Here's what I have: > > > > Results of uname -r: > > 2.6.9-22.0.2.106.unsupportedsmp > > > > Arch: > > X86_64 > > > > If you need more specs on the machine or OS, > please > > let me know. > > > > I downloaded and have been following the asterisk book, and in > > chapter three I followed all the instructions on downloading the > > sources, untarring them, and so forth. > > > > Zaptel compiled without a hitch, as did the rest > of > > the asterisk packages. I > > modified udev, and I restarted the box: ... I did: > > > > /etc/init.d/zaptel start > > > > I get: > > > > Loading zaptel framework: FATAL: Module zaptel > not > > found. > > > > > [FAILED] > > Waiting for zap to come online...Error: missing /dev/zap! > > > > If I do > > > > /sbin/modprobe zaptel > > > > I get: > > FATAL: Module zaptel not found. > > > > If I do > > > > /sbin/modprobe ztdummy > > > > I get: > > > > FATAL: Module ztdummy not found. > > FATAL: Error running install command for ztdummy > > > > Also, if i run: > > > > /etc/init.d/zaptel reload > > > > I get: > > > > Reloading ztcfg: Notice: Configuration file is /etc/zaptel.conf > > line 0: Unable to open master device > '/dev/zap/ctl' > > 1 error(s) detected > > > > > [FAILED] > > > > If I go back to /usr/src/zaptel-1.2.4 and I do > > > > make ztdummy > > > > I get: > > > > cc ztdummy.o -o ztdummy > > > /usr/lib/gcc/x86_64-redhat-linux/3.4.4/../../../../lib64/crt1.o(.text+0x21): > > In > > function `_start': > > : undefined reference to `main' > > ztdummy.o(.text+0xc): In function `ztdummy_timer': > > /usr/src/zaptel-1.2.4/ztdummy.c:154: undefined reference to > > `zt_receive' > > > ztdummy.o(.text+0x18):/usr/src/zaptel-1.2.4/ztdummy.c:155: > > undefined > > reference t > > o `zt_transmit' > > > ztdummy.o(.text+0x1f):/usr/src/zaptel-1.2.4/ztdummy.c:156: > > undefined > > reference t > > o `jiffies' > > ztdummy.o(.text+0x4d): In function `init_module': > > include/linux/slab.h:93: undefined reference to `malloc_sizes' > > ztdummy.o(.text+0x52):include/linux/slab.h:93: > > undefined reference to > > `kmem_cach > > e_alloc' > > ztdummy.o(.text+0x6a): In function `init_module': > > /usr/src/zaptel-1.2.4/ztdummy.c:232: undefined reference to `printk' > > > ztdummy.o(.text+0x197):/usr/src/zaptel-1.2.4/ztdummy.c:192: > > undefined > > reference > > to `zt_register' > > > ztdummy.o(.text+0x1a9):/usr/src/zaptel-1.2.4/ztdummy.c:239: > > undefined > > reference > > to `printk' > > > ztdummy.o(.text+0x1b5):/usr/src/zaptel-1.2.4/ztdummy.c:240: > > undefined > > reference > > to `kfree' > > > ztdummy.o(.text+0x1e2):/usr/src/zaptel-1.2.4/ztdummy.c:261: > > undefined > > reference > > to `jiffies' > > ztdummy.o(.text+0x23d): In function `init_module': > > include/linux/timer.h:87: undefined reference to `__mod_timer' > > ztdummy.o(.text+0x255): In function `init_module': > > /usr/src/zaptel-1.2.4/ztdummy.c:286: undefined reference to `printk' > > ztdummy.o(.text+0x27c): In function > > `cleanup_module': > > /usr/src/zaptel-1.2.4/ztdummy.c:298: undefined reference to > > `del_timer' > > > ztdummy.o(.text+0x288):/usr/src/zaptel-1.2.4/ztdummy.c:303: > > undefined > > reference > > to `zt_unregister' > > > ztdummy.o(.text+0x294):/usr/src/zaptel-1.2.4/ztdummy.c:304: > > undefined > > reference > > to `kfree' > > ztdummy.o(.text+0x39): In function > `ztdummy_timer': > > include/linux/timer.h:87: undefined reference to `__mod_timer' > > ztdummy.o(.text+0x2b0): In function > > `cleanup_module': > > /usr/src/zaptel-1.2.4/ztdummy.c:310: undefined reference to `printk' > > ztdummy.o(__param+0x10): undefined reference to `param_set_int' > > ztdummy.o(__param+0x18): undefined reference to `param_get_int' > > collect2: ld returned 1 exit status > > make: *** [ztdummy] Error 1 > > > > Any ideas? I know I posted things in some wrong order here, but when > > I actually did them as a
RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel
Yes, I did Take care, Sina -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Sunday, March 05, 2006 7:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel did you uncommnet # from before ztdummy ? --- Sina Bahram <[EMAIL PROTECTED]> wrote: > Hi all, > > I hope everyone is doing well. I just joined the list, and I've really > enjoyed all I have read about asterisk so far. > Unfortunately, I'm having a > bit of trouble implementing this thing :). > > By the way ... I did my best to search the forums, and also to use > google extensively, and while I have found pages with people with the > same problem, ... The fix suggested on those sites, didn't work for > me. > > Here's what I have: > > Results of uname -r: > 2.6.9-22.0.2.106.unsupportedsmp > > Arch: > X86_64 > > If you need more specs on the machine or OS, please let me know. > > I downloaded and have been following the asterisk book, and in chapter > three I followed all the instructions on downloading the sources, > untarring them, and so forth. > > Zaptel compiled without a hitch, as did the rest of the asterisk > packages. I modified udev, and I restarted the box: ... I did: > > /etc/init.d/zaptel start > > I get: > > Loading zaptel framework: FATAL: Module zaptel not found. > > [FAILED] > Waiting for zap to come online...Error: missing /dev/zap! > > If I do > > /sbin/modprobe zaptel > > I get: > FATAL: Module zaptel not found. > > If I do > > /sbin/modprobe ztdummy > > I get: > > FATAL: Module ztdummy not found. > FATAL: Error running install command for ztdummy > > Also, if i run: > > /etc/init.d/zaptel reload > > I get: > > Reloading ztcfg: Notice: Configuration file is /etc/zaptel.conf line > 0: Unable to open master device '/dev/zap/ctl' > 1 error(s) detected > > [FAILED] > > If I go back to /usr/src/zaptel-1.2.4 and I do > > make ztdummy > > I get: > > cc ztdummy.o -o ztdummy > /usr/lib/gcc/x86_64-redhat-linux/3.4.4/../../../../lib64/crt1.o(.text+0x21): > In > function `_start': > : undefined reference to `main' > ztdummy.o(.text+0xc): In function `ztdummy_timer': > /usr/src/zaptel-1.2.4/ztdummy.c:154: undefined reference to > `zt_receive' > ztdummy.o(.text+0x18):/usr/src/zaptel-1.2.4/ztdummy.c:155: > undefined > reference t > o `zt_transmit' > ztdummy.o(.text+0x1f):/usr/src/zaptel-1.2.4/ztdummy.c:156: > undefined > reference t > o `jiffies' > ztdummy.o(.text+0x4d): In function `init_module': > include/linux/slab.h:93: undefined reference to `malloc_sizes' > ztdummy.o(.text+0x52):include/linux/slab.h:93: > undefined reference to > `kmem_cach > e_alloc' > ztdummy.o(.text+0x6a): In function `init_module': > /usr/src/zaptel-1.2.4/ztdummy.c:232: undefined reference to `printk' > ztdummy.o(.text+0x197):/usr/src/zaptel-1.2.4/ztdummy.c:192: > undefined > reference > to `zt_register' > ztdummy.o(.text+0x1a9):/usr/src/zaptel-1.2.4/ztdummy.c:239: > undefined > reference > to `printk' > ztdummy.o(.text+0x1b5):/usr/src/zaptel-1.2.4/ztdummy.c:240: > undefined > reference > to `kfree' > ztdummy.o(.text+0x1e2):/usr/src/zaptel-1.2.4/ztdummy.c:261: > undefined > reference > to `jiffies' > ztdummy.o(.text+0x23d): In function `init_module': > include/linux/timer.h:87: undefined reference to `__mod_timer' > ztdummy.o(.text+0x255): In function `init_module': > /usr/src/zaptel-1.2.4/ztdummy.c:286: undefined reference to `printk' > ztdummy.o(.text+0x27c): In function > `cleanup_module': > /usr/src/zaptel-1.2.4/ztdummy.c:298: undefined reference to > `del_timer' > ztdummy.o(.text+0x288):/usr/src/zaptel-1.2.4/ztdummy.c:303: > undefined > reference > to `zt_unregister' > ztdummy.o(.text+0x294):/usr/src/zaptel-1.2.4/ztdummy.c:304: > undefined > reference > to `kfree' > ztdummy.o(.text+0x39): In function `ztdummy_timer': > include/linux/timer.h:87: undefined reference to `__mod_timer' > ztdummy.o(.text+0x2b0): In function > `cleanup_module': > /usr/src/zaptel-1.2.4/ztdummy.c:310: undefined reference to `printk' > ztdummy.o(__param+0x10): undefined reference to `param_set_int' > ztdummy.o(__param+0x18): undefined reference to `param_get_int' > collect2: ld returned 1 exit status > make: *** [ztdummy] Error 1 > > Any ideas? I know I posted things in some wrong order here, but when I > actually did them as a part of the install progress: > I followed the order > layed out in chapter 3 of the book. > > Thanks for any assistance. > > Take care, > Sina > > ___ > --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 > __ Do You Ya
RE: [Asterisk-Users] dtmf tones problem with unicall and E1
|-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Martin Joseph |Sent: Friday, March 03, 2006 1:46 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] dtmf tones problem with unicall and E1 | | |On Mar 3, 2006, at 9:48 AM, Anton Krall wrote: | |> Guys. |> |> I have a te100p with unicall and an E1 and Im having problem |with DTMF |> tones but the weird thing is, I only have problems sending the tones |> to certain phone numbers, anybody seen this behavior? |> |> Asterisk shows on the console the dtmf tone been pressed but |seems the |> other side is not getting them, and this just happens with certain |> phone numbers, not all.. |> |I have seen this through my FXO, when the transmitted volume |is too loud and apparently the audio breaks up at the other |end? This was somewhat speculation on my part, but adjusting |the transmit gain down did seem to resolve it, so that was the proof. | |It's pretty difficult with such a huge variety of different |phone systems and equipment out there to get them all working |acceptably. Of course my biggest issues have been with the |stupid phone system at my wife's workplace! I had to retune |my gains for that system after I thought I was done... | |It's clearly a compromise, and I suppose if the hardware (FXO in my |case) had a GOOD auto gain adjust that might help... | |Marty | |___ |--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 | | Now, what to do when it's a te110p card? E1 in this case .. I don't suppose you can mess with gains ... :( ___ --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] fax receive using TDM400P
On Sat, Feb 25, 2006 at 11:24:36PM +0100, Thomas Artner wrote:
> Am Saturday 25 February 2006 22:59 schrieb Anton Krall:
> > I cant get faxes right now with tdm, something is wrong but, what do I need
> > to have in order to convert from tiff to pdf?
> >
> > I have the mailfax script that invokes tif2ps and ps2pdf but pages come out
> > blank..
> >
>
>
> I do the following:
>
> exten => fax,1,Set(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID})
> exten => fax,2,rxfax(${FAXFILE})
> exten => fax,3,system(tiff2pdf ${FAXFILE} > ${FAXFILE}.pdf)
> exten => fax,4,system(mpack -s "received Fax" -c application/octet-stream
> ${FAXFILE}.pdf [EMAIL PROTECTED])
>
If you give the mime type explicitly, give a correct one, so the user
can know what program to use.
For a PDF file, use: application/pdf
This will make properly-configured mailers launch a PDF reader.
Example command-line mailers that can give PDF files a proper mime type:
mutt -a ${FAXFILE}.pdf -s "your fax" [EMAIL PROTECTED] http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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[Asterisk-Users] re: Sixtel Services
Hi, Companies like DIDx and Sixtel, when they state DIDs at $XX.XX per month and $XX.XX per minute/monthly, do these companies provide inbound and outbound routing of calls, or are these rates strictly for inbound Call routing of DIDs? Thanks. ___ --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] Dialplan - strip IDD prefix and insert another
SellVoIP appears to follow a US dialplan. A US number is dialled as 1NXXNXX whereas an international (to the US) number is dialled as 011X. Frankly, I didn't ask whether international numbers like Barbados where the code remains as 1 but are international (to the US) need the 011 or can be dialled directly but that's not really my concern. I've assumed they don't. Most of the world uses 00 as the internation prefix code, therefore I have to ask: How can I "strip" the 00 and insert 011 in one entry in the dialplan. I'm stripping the 00 and passing the rest of the numbers for numbers dialled as 001X. (as in: 00|1XX.) but in case of numbers out of the US, how would I insert the 011 ? BEGIN:VCARD VERSION:2.1 N:Tarzi;AbdelRahman el FN:AbdelRahman el Tarzi ORG:Arab Banking Corporation;Proprietary Investment TITLE:Structured Credit Derivatives NOTE;ENCODING=QUOTED-PRINTABLE:Fax: +973 39 33 27 69=0D=0AContacts in Egypt: =0D=0ACell: +20(10) 1236700= =0D=0ACairo: Residence: +20 (2) 4028860=0D=0AMarina: Residence: +20 (46) 406= 2197 (temp unavailable)=0D=0AZomorroda: Residence: +20 (3) 5210765=0D=0A TEL;WORK;VOICE:+973 1754 3700 TEL;HOME;VOICE:+973 17 69 80 24 TEL;CELL;VOICE:+973 39 68 57 00 TEL;WORK;FAX:+973 1753 1427 ADR;WORK:;3rd floor, ABC Building;P.O. BOX 5698;Manama;;;Bahrain LABEL;WORK;ENCODING=QUOTED-PRINTABLE:3rd floor, ABC Building=0D=0AP.O. BOX 5698=0D=0AManama=0D=0ABahrain ADR;HOME;ENCODING=QUOTED-PRINTABLE:;;House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain;Manama;;= ;Bahrain LABEL;HOME;ENCODING=QUOTED-PRINTABLE:House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain=0D=0AManam= a=0D=0ABahrain X-WAB-GENDER:2 URL;WORK:www.arabbanking.com BDAY:20050123 KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZWMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTEzNTFaFw0wNjExMTEwOTEz NTFaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAeWFob28u Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQCvGOn8FwM/UUm7OYMdFZYn+hUrmDYo ARJGJvFDu7lnbrT/v3tf1zRpOULT8yN2PXtSUmsxlvYX2SCJ8PggECGGbyJEkd8bHmPJEi7g FHNs9h3ps7SJ+gQFkqa0soxegfHgQzrjrOGXNI1dMCKaYc6a2dSWRUBj4C1ii1dHYs7jmQID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQHlhaG9vLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAC9Tm59BZjKmw61xcYa4yXhPSqfkXTJy6eAVX4LSwM1gkRbV6HWZ HjQBmEhTkfrAF01xeKrDRh6vJIYGjSuPJRVmCN2+BA/UuNnK3EQOI+mwuku8KQzDAFXpJHhe +J5626T7NiuADtT2F0L3tLoFf8vvLcyTzvCHU+y6E2Danaak KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZXMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTE5MDRaFw0wNjExMTEwOTE5 MDRaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAZ21haWwu Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQDASKRiH2YqhCqPF3HDlPCdtHZb78Pn Z4S/qzgdLVdzeE1b2Ddd4gl+FkQw2IS4Q+3XSwsGyh9wY6irNb+nIrr5Gs9+JmpQTSPjQp72 trLvD+PvFetwQMotRODVsgxHIpgcTFBjpMZ4P24NeAGRBNzfPjwqx3gfscd10fWtiXGo8wID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQGdtYWlsLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAAZ2rAEswRkNEgiMcy3enKlTcQ9QiIFeQP5bq7iXDUkbhtcZHDdi ol+HaN6QyO2ZUCYbuK1d12VD92QpZuRxw0lS7K7qWU7aF5gabpnEjl1KQ0ujr+gEcV2ogvZY 2F4SZ7H9uF0c06/NT5TpoFyok3wJ/jZXJhRAbR/Eye678OCq KEY;X509;ENCODING=BASE64: MIICfDCCAeWgAwIBAgIDD80vMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMDYyMTMzMzVaFw0wNjExMDYyMTMz MzVaMG8xDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxJTAjBgkqhkiG9w0BCQEWFmFydGFyemlAYmF0ZWxj by5jb20uYmgwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAK+koXkgs50JRrsTV4tj2QS7 uZ05+iKe/lhkdv56a6oEUcw4tO03rGMcB+ocWwfmmIbZ1n5p8dRjybsZMI5zEnRsf/KeQLl3 1wBPYoKzVDQrulNMGh8FmhK8uWsW1FZSKJkbxZWjcI2fkbDLmQuvWBUdlgiOFOLp08m9bMvf ZpCfAgMBAAGjMzAxMCEGA1UdEQQaMBiBFmFydGFyemlAYmF0ZWxjby5jb20uYmgwDAYDVR0T AQH/BAIwADANBgkqhkiG9w0BAQQFAAOBgQA/TNRreOLNx7d1f7H9vfrnlTRuftVHVL4f6h6X u2Od18TDDP6/iUuiTtcMQfOOwiBBxjkgdupsDi4q8FrOseWu5ylM9hNg+1mtjSQT00CL6n4A CIh94LiywiMeJmxzKLuihUxyQu2aRFksaQS4unmENCZ23a+xB4DHuTD9V3FcAy== EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;PREF;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] REV:20060305T155750Z 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
[Asterisk-Users] Inserting access codes as prefixes to CID
When I receive a call from fwd, I'd like to insert a prefix prior to the caller ID - 1) to be able to look it up in a database of identified numbers and 2) for the receiver to be able to dial it back. So what I need is to identify the DID and based on that, insert the prefix. Any pointers to documentation would be appreciated BEGIN:VCARD VERSION:2.1 N:Tarzi;AbdelRahman el FN:AbdelRahman el Tarzi ORG:Arab Banking Corporation;Proprietary Investment TITLE:Structured Credit Derivatives NOTE;ENCODING=QUOTED-PRINTABLE:Fax: +973 39 33 27 69=0D=0AContacts in Egypt: =0D=0ACell: +20(10) 1236700= =0D=0ACairo: Residence: +20 (2) 4028860=0D=0AMarina: Residence: +20 (46) 406= 2197 (temp unavailable)=0D=0AZomorroda: Residence: +20 (3) 5210765=0D=0A TEL;WORK;VOICE:+973 1754 3700 TEL;HOME;VOICE:+973 17 69 80 24 TEL;CELL;VOICE:+973 39 68 57 00 TEL;WORK;FAX:+973 1753 1427 ADR;WORK:;3rd floor, ABC Building;P.O. BOX 5698;Manama;;;Bahrain LABEL;WORK;ENCODING=QUOTED-PRINTABLE:3rd floor, ABC Building=0D=0AP.O. BOX 5698=0D=0AManama=0D=0ABahrain ADR;HOME;ENCODING=QUOTED-PRINTABLE:;;House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain;Manama;;= ;Bahrain LABEL;HOME;ENCODING=QUOTED-PRINTABLE:House 758=0D=0ARoad 2033=0D=0ABlock 520 Barbar=0D=0A=0D=0ABahrain=0D=0AManam= a=0D=0ABahrain X-WAB-GENDER:2 URL;WORK:www.arabbanking.com BDAY:20050123 KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZWMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTEzNTFaFw0wNjExMTEwOTEz NTFaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAeWFob28u Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQCvGOn8FwM/UUm7OYMdFZYn+hUrmDYo ARJGJvFDu7lnbrT/v3tf1zRpOULT8yN2PXtSUmsxlvYX2SCJ8PggECGGbyJEkd8bHmPJEi7g FHNs9h3ps7SJ+gQFkqa0soxegfHgQzrjrOGXNI1dMCKaYc6a2dSWRUBj4C1ii1dHYs7jmQID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQHlhaG9vLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAC9Tm59BZjKmw61xcYa4yXhPSqfkXTJy6eAVX4LSwM1gkRbV6HWZ HjQBmEhTkfrAF01xeKrDRh6vJIYGjSuPJRVmCN2+BA/UuNnK3EQOI+mwuku8KQzDAFXpJHhe +J5626T7NiuADtT2F0L3tLoFf8vvLcyTzvCHU+y6E2Danaak KEY;X509;ENCODING=BASE64: MIICcjCCAdugAwIBAgIDD9ZXMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMTEwOTE5MDRaFw0wNjExMTEwOTE5 MDRaMGoxDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxIDAeBgkqhkiG9w0BCQEWEWFydGFyemlAZ21haWwu Y29tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQDASKRiH2YqhCqPF3HDlPCdtHZb78Pn Z4S/qzgdLVdzeE1b2Ddd4gl+FkQw2IS4Q+3XSwsGyh9wY6irNb+nIrr5Gs9+JmpQTSPjQp72 trLvD+PvFetwQMotRODVsgxHIpgcTFBjpMZ4P24NeAGRBNzfPjwqx3gfscd10fWtiXGo8wID AQABoy4wLDAcBgNVHREEFTATgRFhcnRhcnppQGdtYWlsLmNvbTAMBgNVHRMBAf8EAjAAMA0G CSqGSIb3DQEBBAUAA4GBAAZ2rAEswRkNEgiMcy3enKlTcQ9QiIFeQP5bq7iXDUkbhtcZHDdi ol+HaN6QyO2ZUCYbuK1d12VD92QpZuRxw0lS7K7qWU7aF5gabpnEjl1KQ0ujr+gEcV2ogvZY 2F4SZ7H9uF0c06/NT5TpoFyok3wJ/jZXJhRAbR/Eye678OCq KEY;X509;ENCODING=BASE64: MIICfDCCAeWgAwIBAgIDD80vMA0GCSqGSIb3DQEBBAUAMGIxCzAJBgNVBAYTAlpBMSUwIwYD VQQKExxUaGF3dGUgQ29uc3VsdGluZyAoUHR5KSBMdGQuMSwwKgYDVQQDEyNUaGF3dGUgUGVy c29uYWwgRnJlZW1haWwgSXNzdWluZyBDQTAeFw0wNTExMDYyMTMzMzVaFw0wNjExMDYyMTMz MzVaMG8xDjAMBgNVBAQTBVRhcnppMRcwFQYDVQQqEw5BYmRlbFJhaG1hbiBFbDEdMBsGA1UE AxMUQWJkZWxSYWhtYW4gRWwgVGFyemkxJTAjBgkqhkiG9w0BCQEWFmFydGFyemlAYmF0ZWxj by5jb20uYmgwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAK+koXkgs50JRrsTV4tj2QS7 uZ05+iKe/lhkdv56a6oEUcw4tO03rGMcB+ocWwfmmIbZ1n5p8dRjybsZMI5zEnRsf/KeQLl3 1wBPYoKzVDQrulNMGh8FmhK8uWsW1FZSKJkbxZWjcI2fkbDLmQuvWBUdlgiOFOLp08m9bMvf ZpCfAgMBAAGjMzAxMCEGA1UdEQQaMBiBFmFydGFyemlAYmF0ZWxjby5jb20uYmgwDAYDVR0T AQH/BAIwADANBgkqhkiG9w0BAQQFAAOBgQA/TNRreOLNx7d1f7H9vfrnlTRuftVHVL4f6h6X u2Od18TDDP6/iUuiTtcMQfOOwiBBxjkgdupsDi4q8FrOseWu5ylM9hNg+1mtjSQT00CL6n4A CIh94LiywiMeJmxzKLuihUxyQu2aRFksaQS4unmENCZ23a+xB4DHuTD9V3FcAw== EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;PREF;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] EMAIL;INTERNET:[EMAIL PROTECTED] REV:20060305T155747Z 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] really need help with outgoing calls..PSTN errors
I have to wait until Monday to test but I will make that change.thanks Rich Adamson <[EMAIL PROTECTED]> wrote: Might take a close look at group => 1 in your zapata.conf file. Thatshould be group=1.Someone mentioned adding "w" into your outbound calls, like: exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1})Did you try that in each of your Dial strings?> In our area code(703), and I am not sure if it is like this in other places, we are required to dial the area code even if we dial local> numbers . That is what these lines are for:> > exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> exten => _9XX,2,Congestion()> exten => _9XX,102,Congestion()> > Any other options?> >
> Mark Hulber <[EMAIL PROTECTED]> wrote:> > Have you tried dialing an 800 number? Does that work? This extension:> > exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > seems to be missing one X since it's only 10 digits long. Your PSTN> probably requires a 1 to be dialed also. On the other hand,> > exten => _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > you should probably be matching this extension instead although you> won't be able to match anywhere that has an area code that starts with> an 8 or 9. (905, 916, 914 as a few examples).> > MARK.> > sdgesa gaeharth wrote:> > I cant seem to get outgoing calls to be placed properly .. No matter> > what I try I get an error from the PSTN company saying that the "call> > can not be comple
ted as
dialed" or "you need to dial a one..." The> > asterisk debugging seems to show the correct number being dialed out> > of the zap interface... the "9" is being stripped and there is a "1"> > where it is supposed to be. I am thinking it is a problem between the> > zap interface and the PSTN.> >> > thanks> >> > extensions.conf> > [general]> > static=yes> > writeprotect=no> > autofallthrough=yes> > clearglobalvars=no> > priorityjumping=no> > [globals]> > ATTENDANT=1001> > OUTBOUNDTRUNK=ZAP/g1> > [extentions]> > exten => _10XX,1,Ringing> > exten => _10XX,2,Dial(SIP/${EXTEN},20)> > exten => _10XX,3,Answer> > exten => _10XX,4,VoiceMail([EMAIL PROTECTED]>
>
)> > exten => _10XX,5,Hangup> > [voicemail]> > exten => _910XX,1,Wait(1)> > exten => _910XX,2,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])> > [local]> > include => extentions> > include => voicemail> > [incoming]> > exten => s,1,Answer> > exten => s,n,Wait(2)> > exten => s,n,Set(TIMEOUT(response)=15)> > exten => s,n,Background(company-intro)> > exten => s,n,WaitExten()> > exten => s,n,Playback(vm-goodbye)> > exten => s,n,Hangup()> > exten => 0,1,Dial(SIP/${ATTENDANT},20)> > exten => 1,1,Directory(voicemail,extentions,f)> > exten => 2,1,Directory(voicemail,extentions)> > exten => 1234,1,Playback(abandon-all-hope)> > include => extentions> > exten
=>
i,1,Playback(vm-goodbye)> > exten => i,2,Hangup()> > exten => t,1,Playback(vm-goodbye)> > exten => t,2,Hangup()> > [outbound]> > ignorepat => 9> > exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > exten => _9XX,2,Congestion()> > exten => _9XX,102,Congestion()> > exten => _91800NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > exten => _91800NXX,2,Congestion()> > exten => _91800NXX,102,Congestion()> > exten => _91888NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > exten => _91888NXX,2,Congestion()> > exten => _91888NXX,102,Congestion()> > exten => _91877NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > exten => _91877NXX,2,Congestion()> > exten =>
_91877NXX,102,Congestion()> > exten => _91866NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > exten => _91866NXX,2,Congestion()> > exten => _91866NXX,102,Congestion()> > exten => _91900NXX,1,Congestion()> > exten => _91976NXX,1,Congestion()> > exten => _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})> > exten => _91[1234567]XXNXX,2,Congestion()> > exten => _91[1234567]XXNXX,102,Congestion()> > exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)> > exten => 9411,1,Dial(${OUTBOUNDTRUNK}/411)> > exten => 0,1,Dial(${OUTBOUNDTRUNK}/0)> >> > [local-access]> > include => local> > include => outbound> >> > zapata.conf:> > [channels]> > group => 1>
>
language=en> > context=incoming> > signalling=fxs_ks> > switchtype=national> > usecallerid=yes> > hidecallerid=no> > callwaiting=yes> > callerid => "Dulles Micro, LLC" <703 450 5000>> > usecallingpres=yes> > callwaitingcallerid=yes> > threewaycalling=yes> > transfer=yes> > canpark=yes> > cancallforward=yes> > callreturn=yes> > echocancel=yes> > echocancelwhenbridged=yes>
Re: [Asterisk-Users] Re: uniqueid
You need to compile asterisk-addons with CFLAGS+=-DMYSQL_LOGUNIQUEIDcheck: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20mysql On 3/5/06, FaberK <[EMAIL PROTECTED]> wrote: News!I've just replaced the cdr_addon_mysql.so with the old one, and it start to work properly!So I can suppose a bug into that module.I'll check the old cdr_addon_mysql.c and see difference of code, if any. Thanks.2006/3/5, FaberK <[EMAIL PROTECTED]>: Hi folks,I've just updated my * and I noticed that from the update the uniqueid field into mysql, is not written and ASTPP do not charge the calls.I got an eye to cdr_mysql.c and I found that at line 212, into one insert query, uniqueid is missing. But I can be wrong.In any case, somebody got same problem?Any suggestions?Thanks to all.-- .:FaberK:. -- .:FaberK:. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Carlo TaguinodLinux Registered User #283313 (counter.li.org) Brainbench Transcript #4381927(www.brainbench.com) ___ --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] really need help with outgoing calls..PSTN errors
Might take a close look at group => 1 in your zapata.conf file. That
should be group=1.
Someone mentioned adding "w" into your outbound calls, like:
exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1})
Did you try that in each of your Dial strings?
> In our area code(703), and I am not sure if it is like this in other places,
> we are required to
dial the area code even if we dial local
> numbers . That is what these lines are for:
>
> exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> exten => _9XX,2,Congestion()
> exten => _9XX,102,Congestion()
>
> Any other options?
>
>
> Mark Hulber <[EMAIL PROTECTED]> wrote:
>
> Have you tried dialing an 800 number? Does that work? This extension:
>
> exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
>
> seems to be missing one X since it's only 10 digits long. Your PSTN
> probably requires a 1 to be dialed also. On the other hand,
>
> exten => _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
>
> you should probably be matching this extension instead although you
> won't be able to match anywhere that has an area code that starts with
> an 8 or 9. (905, 916, 914 as a few examples).
>
> MARK.
>
> sdgesa gaeharth wrote:
> > I cant seem to get outgoing calls to be placed properly .. No matter
> > what I try I get an error from the PSTN company saying that the "call
> > can not be completed as dialed" or "you need to dial a one..." The
> > asterisk debugging seems to show the correct number being dialed out
> > of the zap interface... the "9" is being stripped and there is a "1"
> > where it is supposed to be. I am thinking it is a problem between the
> > zap interface and the PSTN.
> >
> > thanks
> >
> > extensions.conf
> > [general]
> > static=yes
> > writeprotect=no
> > autofallthrough=yes
> > clearglobalvars=no
> > priorityjumping=no
> > [globals]
> > ATTENDANT=1001
> > OUTBOUNDTRUNK=ZAP/g1
> > [extentions]
> > exten => _10XX,1,Ringing
> > exten => _10XX,2,Dial(SIP/${EXTEN},20)
> > exten => _10XX,3,Answer
> > exten => _10XX,4,VoiceMail([EMAIL PROTECTED]
> > )
> > exten => _10XX,5,Hangup
> > [voicemail]
> > exten => _910XX,1,Wait(1)
> > exten => _910XX,2,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
> > [local]
> > include => extentions
> > include => voicemail
> > [incoming]
> > exten => s,1,Answer
> > exten => s,n,Wait(2)
> > exten => s,n,Set(TIMEOUT(response)=15)
> > exten => s,n,Background(company-intro)
> > exten => s,n,WaitExten()
> > exten => s,n,Playback(vm-goodbye)
> > exten => s,n,Hangup()
> > exten => 0,1,Dial(SIP/${ATTENDANT},20)
> > exten => 1,1,Directory(voicemail,extentions,f)
> > exten => 2,1,Directory(voicemail,extentions)
> > exten => 1234,1,Playback(abandon-all-hope)
> > include => extentions
> > exten => i,1,Playback(vm-goodbye)
> > exten => i,2,Hangup()
> > exten => t,1,Playback(vm-goodbye)
> > exten => t,2,Hangup()
> > [outbound]
> > ignorepat => 9
> > exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> > exten => _9XX,2,Congestion()
> > exten => _9XX,102,Congestion()
> > exten => _91800NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> > exten => _91800NXX,2,Congestion()
> > exten => _91800NXX,102,Congestion()
> > exten => _91888NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> > exten => _91888NXX,2,Congestion()
> > exten => _91888NXX,102,Congestion()
> > exten => _91877NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> > exten => _91877NXX,2,Congestion()
> > exten => _91877NXX,102,Congestion()
> > exten => _91866NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> > exten => _91866NXX,2,Congestion()
> > exten => _91866NXX,102,Congestion()
> > exten => _91900NXX,1,Congestion()
> > exten => _91976NXX,1,Congestion()
> > exten => _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> > exten => _91[1234567]XXNXX,2,Congestion()
> > exten => _91[1234567]XXNXX,102,Congestion()
> > exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)
> > exten => 9411,1,Dial(${OUTBOUNDTRUNK}/411)
> > exten => 0,1,Dial(${OUTBOUNDTRUNK}/0)
> >
> > [local-access]
> > include => local
> > include => outbound
> >
> > zapata.conf:
> > [channels]
> > group => 1
> > language=en
> > context=incoming
> > signalling=fxs_ks
> > switchtype=national
> > usecallerid=yes
> > hidecallerid=no
> > callwaiting=yes
> > callerid => "Dulles Micro, LLC" <703 450 5000>
> > usecallingpres=yes
> > callwaitingcallerid=yes
> > threewaycalling=yes
> > transfer=yes
> > canpark=yes
> > cancallforward=yes
> > callreturn=yes
> > echocancel=yes
> > echocancelwhe
Re: [Asterisk-Users] 160 analogue phones..
Conrad, i would go with following solution: 1. 6 sets of Audio Codes of 24 FXS ports conected by SIP accounts to the system. the type is MP 124. then you open the conector on the initial MDF and then the users have the same phone on their table 2. one dual Xeon system (or even stronger - 2 Dual Core system). such a configuration can take 60 calls at g711. 3. 16 IP phones for the medium up users i hope i helped you in a way. Mickey On 3/1/06, Conrad Wood <[EMAIL PROTECTED]> wrote: Does anyone have any recommendations on how to connect 160 analoguephones to an asterisk PBX?Background information: A client wishes to replace their current PBX with a new VoIP system.Currently they have 2 PRIs.I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raideddrives. These drives will be mounted only read-only to recover gracefully from power-cycles. I am considering 2 ISDNGuards in front ofthe machines.More to the point: The client has 160 existing analogue telephones whichthey don't really want to change right now, because a) they are very cheap b) the users don't need to re-train.I have thought of Rhino Channelbanks, but then realised I need to use 7of them and connect each with a T1. I don't really want to run 7 T1 +the 2 PRIs into one asterisk box for performance reasons. Ideally, several 48-Port SIP->FXS channelbank woulds be ideal Iguess ;-). Does such thing exist? Or how do others do this?Conrad___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Program Buttons on Cisco 79xx Phones
> >> We're still waiting for a SIP-enabled 7970... > >> The newer model phones (7941g/ge, 7961g) are sccp-only. Seems a step > >> backwards to me. > > why? > > If cisco really is moving towards SIP as claimed earlier, then releasing > new phones which are sccp-only is a step backwards from that goal. > > If cisco really is moving towards SIP as claimed earlier, then cisco > should release SIP images for 7970, as they did with 7960 and 7940. > > > I had my phones running on SIP, got chan-sccp and started > > experimenting with it. > > All my phones are running SCCP now. The phones respond > > faster, you have more options etc. > > Of course it would be nice if they offer SIP so people have > > a choice, but I really think the chan_sccp is the way to > > have these phones work. > > This only means sccp is currently better for cisco phones, it doesn't mean > sccp is a better protocol. > > sccp and asterisk has some err.. real annoying bugs at the moment, where > ciscos running SIP don't have these problems. > > Given a choice I'd run SIP, if only to have the phones able to talk to > each other and gateways if the PBX dies for any reason. sccp can't do > that -- if you lose the PBX you totally lose all functionality on all > your sccp phones. Cisco has a very long track history of supporting "standards". However, I'm sure they balance that against their interpretation of how that support might impact sales of other products, and against the development time necessary to get there. Part of that decision might even relate to the maturity of their sip code in CM. I'd have to bet they will have better sip implementations for all phones in the near future, particularily when competitive products become more advanced/stable. Since the v7.5 sip code had a significant number of problems (compared to all other firmware versions released in the last two years), it would appear that might be a "leading indicator" of some significant development efforts that we've just beginning to see. ___ --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] Re: uniqueid
News!I've just replaced the cdr_addon_mysql.so with the old one, and it start to work properly!So I can suppose a bug into that module.I'll check the old cdr_addon_mysql.c and see difference of code, if any. Thanks.2006/3/5, FaberK <[EMAIL PROTECTED]>: Hi folks,I've just updated my * and I noticed that from the update the uniqueid field into mysql, is not written and ASTPP do not charge the calls.I got an eye to cdr_mysql.c and I found that at line 212, into one insert query, uniqueid is missing. But I can be wrong.In any case, somebody got same problem?Any suggestions?Thanks to all.-- .:FaberK:. -- .:FaberK:. ___ --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] 20 seconds til voice transmission starts
> I'm experiencing a strange problem with my Asterisk. I hope you can help: > > Asterisk is running at my company behind NAT. Ports 5060 and 1-2 > are being forwarded to it. I have put the router's external IP-address > into externip in sip.conf. At home I'm using an AVM FritzBox Fon WLAN > 7050 which is registered with the Asterisk at my company. > > When I try to call Asterisk (or a phone connected to the attached > legacy-pbx) from home, it's ringing normally and I can hear my opposite. > But it takes about 20 seconds until my opposite hears me! When I call > the same number again staight after, everything is working fine from the > beginning. Also, calls from the company to my home are working perfectly. > > I'm greateful for any tips! One way to identify the issue is to run ethereal to see what's happening with the udp ports. If that doesn't provide a clue, then run asterisk with additional levels of debug/verboseness. ___ --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] uniqueid
Hi folks,I've just updated my * and I noticed that from the update the uniqueid field into mysql, is not written and ASTPP do not charge the calls.I got an eye to cdr_mysql.c and I found that at line 212, into one insert query, uniqueid is missing. But I can be wrong.In any case, somebody got same problem?Any suggestions?Thanks to all.-- .:FaberK:. ___ --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] Realtime Content on LCD Display
Hello, Anyone knows a way to show real-time content from a DB into the LCD display of an IP phone, like any 79xx? If someone knows which phone is capable of doing and how, like using XML files, please advise. Regards, Max Glucksmann e-mail: [EMAIL PROTECTED] Web: http://www.comtel-networks.com Venezuela Teléfono: (0500) MAXITEL ext. 1011001 Fax: (0212) 953-0769 USA Phone: 1 (877) 467-2877 ext. 1011001 Fax: (954) 671-6800 BEGIN:VCARD VERSION:2.1 N:Glucksmann;Max FN:Max Glucksmann (Fax del trabajo) ORG:ComTel Networks, Corp. TITLE:Director TEL;WORK;VOICE:+1 (877) 467-2877 TEL;HOME;VOICE:+58 (500) MAXITEL (629-4835) TEL;CELL;VOICE:+58 (414) 250-0909 TEL;WORK;FAX:+1 (954) 671-6800 TEL;HOME;FAX:+58 (212) 285-3320 ADR;WORK:;;Aerocav 1614, PO Box 25304;Miami;FL.;33102-5304;Estados Unidos de América LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Aerocav 1614, PO Box 25304=0D=0AMiami, FL. 33102-5304=0D=0AEstados Unidos de= Am=E9rica EMAIL;PREF;FAX:Max Glucksmann ([EMAIL PROTECTED])@+1 (954) 671-6800 REV:20051212T222729Z END:VCARD BEGIN:VCARD VERSION:2.1 N:Glucksmann;Max FN:Max Glucksmann (Fax del trabajo) ORG:ComTel Networks, Corp. TITLE:Director TEL;WORK;VOICE:+1 (877) 467-2877 TEL;HOME;VOICE:+58 (500) MAXITEL (629-4835) TEL;CELL;VOICE:+58 (414) 250-0909 TEL;WORK;FAX:+1 (954) 671-6800 TEL;HOME;FAX:+58 (212) 285-3320 ADR;WORK:;;Aerocav 1614, PO Box 25304;Miami;FL.;33102-5304;Estados Unidos de América LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Aerocav 1614, PO Box 25304=0D=0AMiami, FL. 33102-5304=0D=0AEstados Unidos de= Am=E9rica EMAIL;PREF;FAX:Max Glucksmann ([EMAIL PROTECTED])@+1 (954) 671-6800 REV:20051212T222729Z 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] Auto dial feature
Kevin Smith wrote:
Hey everyone,
We have a special mail box for certain customers when we are out of
the office. Basically they enter a pin number and if it is valid they
leave a message and it notifies the on call techs. My question is
regarding externnotify for the voice mail.conf. If I enabled that and
set up a call file, will it do it for every voice mail box I have on
the system?
Yes
Is there a way I can limit it to just the one voice mail box on the
system?
This is what I do:
Create a database entry for that extension, I call it vmcallback. The
entries can either be YES or NO. At every point in your dial plan you
need to check for that entry against the extension that voice mail is
being left for. If the value is YES, then run the script that copies
the .call file to the outgoing.
[macro-sip.extensions]
exten => s,1,Set(CALLBACK=${DB(vmcallback/${ARG1})})
exten => s,2,NoOP(${CALLBACK})
exten => s,3,SetMusicOnHold(epi-cd)
exten => s,4,Dial(SIP/${ARG1},28,t)
exten => s,5,NoOP(Dial Status: ${DIALSTATUS})
exten => s,6,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten => s,7,Goto(s-${DIALSTATUS},1)
exten => s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten => s-NOANSWER,1,GotoIf($["${CALLBACK}" =
"YES"]?s-NOANSWER,2:s-NOANSWER,3)
exten => s-NOANSWER,2,System(/usr/local/bin/vm-callout.sh ${ARG1})
exten => s-NOANSWER,3,Voicemail([EMAIL PROTECTED])
exten => s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten => s-CANCEL,1,Congestion()
exten => h,1,NoOP(Hungup)
My callout script copies a pre-created .call file, sets the date 5
minutes into the future, copies it to the outgoing directory (While
preserving the time stamps). When the 5 minutes have passed, Asterisk
acts on it. Script contents below:
#!/bin/sh
cd /usr/local/bin
/bin/touch /usr/local/bin/$1.call
/bin/touch -r /usr/local/bin/$1.call -m -F 300 /usr/local/bin/$1.call
cp --preserve=timestamps /usr/local/bin/$1.call
/var/spool/asterisk/outgoing/
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety."
___
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Re: ***SPAM*** Re: [Asterisk-Users] D-Link DVG-1402S
On Sun, 05 Mar 2006 19:56:13 +0800 Stephen Arulraj <[EMAIL PROTECTED]> wrote: > Come on.! Don't tell me no one has ever had a problem on this model > with asterisk? Live it up guys... and make a few comments maybe you would get more answers if you wouldn't steal a thread, but would create your own. For me it is not clear how your message belongs to the thread "No audio on PRI". Gerald ___ --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] 160 analogue phones..
I would look at the cost of the channle banks vs. selling the analog phones and getting very basic voip hardphones. --- Conrad Wood <[EMAIL PROTECTED]> wrote: > Does anyone have any recommendations on how to > connect 160 analogue > phones to an asterisk PBX? > > Background information: > A client wishes to replace their current PBX with a > new VoIP system. > Currently they have 2 PRIs. > I intent to set up 2 asterisk PBXs with Debian > GNU/Linux on raided > drives. These drives will be mounted only read-only > to recover > gracefully from power-cycles. I am considering 2 > ISDNGuards in front of > the machines. > More to the point: The client has 160 existing > analogue telephones which > they don't really want to change right now, because > a) they are very > cheap b) the users don't need to re-train. > > I have thought of Rhino Channelbanks, but then > realised I need to use 7 > of them and connect each with a T1. I don't really > want to run 7 T1 + > the 2 PRIs into one asterisk box for performance > reasons. > > Ideally, several 48-Port SIP->FXS channelbank woulds > be ideal I > guess ;-). Does such thing exist? Or how do others > do this? > > Conrad > > > ___ > --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 > __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Preferred editor(s) dialplan coding?
VI as well but sometimes I use the editor built into WinSCP. Thanks, Steve Totaro > -Original Message- > From: C F [mailto:[EMAIL PROTECTED] > Sent: Saturday, March 04, 2006 9:22 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Preferred editor(s) dialplan coding? > > vi here > > On 3/4/06, JP Carballo <[EMAIL PROTECTED]> wrote: > > > > Bill Gibbs wrote: > > > > >Vim for everything > > > > > >-Original Message- > > >From: [EMAIL PROTECTED] > > >[mailto:[EMAIL PROTECTED] On Behalf Of Pete > > >Barnwell > > >Sent: Friday, March 03, 2006 7:39 PM > > >To: Asterisk Users Mailing List - Non-Commercial Discussion > > >Subject: Re: [Asterisk-Users] Preferred editor(s) dialplan coding? > > > > > >Emacs... > > > > > >On Sat, 2006-03-04 at 01:35 +0100, adibar wrote: > > > > > > >Vim forever ;-) > > > > > > > >On Fri, Mar 03, 2006 at 03:06:02PM -0500, S McGowan wrote: > > > > > > > > emacs for me :) > > > > -- > > JP Carballo > > > > http://www.netfone2x.com > > Bringing the world closer. > > > > It might look like I'm doing nothing, but at the cellular level, I'm > really quite busy. > > > > ___ > > --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
Re: [Asterisk-Users] How to route incoming calls to different contexts?
hi Zach, i would use GOTOIF to forward the DID from within the [incoming] context to the other context. i would try : exten => gotoif($[did]=DID1,goto did1|s|1,) exten => gotoif($[did]=DID2,goto did2|s|1,) On 3/4/06, Zach A <[EMAIL PROTECTED]> wrote: Both DIDs are SIP and from the same provider. Format of registration islike this:sip.conf [general]bindaddr=xxx.xxx.xxx.xxxport=5060context=incomingdisallow=allallow=g726allow=ulawallow=alawallow=gsmdtmfmode=rfc2833canreinvite=no ; required for incoming calls to ring extensions insecure=invite ; outgoing call not working without thistos=0x18nat=yesregister=DID1:[EMAIL PROTECTED]register= DID2:[EMAIL PROTECTED][DID1]username=DID1type=peersecret=1234host=xxx.xxx.xxx.xxxfromuser=DID1[DID2]username=DID2type=peersecret=1234host=xxx.xxx.xxx.xxxfromuser=DID2 Now both DIDs are sent to context [incoming] which is the defaultcontext for SIP. If I add context=incoming2 under any DID section, itdoesn't go to that context and still go to the default context. How can I direct DID2 to [incoming2] context?Zach A-Original Message-From: Joseph Tanner [mailto:[EMAIL PROTECTED]]Sent: Friday, March 03, 2006 7:18 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] How to route incoming calls todifferentcontexts?First, tell us if it's sip, iax, or zap. Then tell us what provider (most will use the same general config, but some like ipkall arespecial and a bit tricky).joseph TannerOn 3/3/06, Zach A <[EMAIL PROTECTED]> wrote: Hi everybody, It should be a simple thing to do but I don't know how to do it. Now Ihave> 2 DIDs and I want one of them go to [context1] and other one to go to > [context2]. How can I achieve this. Thanks, Zach A> ___> --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Problem compiling ztdummy on centos 4, 2.6 kernel
make linux26 make install worked for me thanks --- Dovid Bender <[EMAIL PROTECTED]> wrote: > did you uncommnet # from before ztdummy ? > > --- Sina Bahram <[EMAIL PROTECTED]> wrote: > > > Hi all, > > > > I hope everyone is doing well. I just joined the > > list, and I've really > > enjoyed all I have read about asterisk so far. > > Unfortunately, I'm having a > > bit of trouble implementing this thing :). > > > > By the way ... I did my best to search the forums, > > and also to use google > > extensively, and while I have found pages with > > people with the same problem, > > ... The fix suggested on those sites, didn't work > > for me. > > > > Here's what I have: > > > > Results of uname -r: > > 2.6.9-22.0.2.106.unsupportedsmp > > > > Arch: > > X86_64 > > > > If you need more specs on the machine or OS, > please > > let me know. > > > > I downloaded and have been following the asterisk > > book, and in chapter three > > I followed all the instructions on downloading the > > sources, untarring them, > > and so forth. > > > > Zaptel compiled without a hitch, as did the rest > of > > the asterisk packages. I > > modified udev, and I restarted the box: ... I did: > > > > /etc/init.d/zaptel start > > > > I get: > > > > Loading zaptel framework: FATAL: Module zaptel > not > > found. > > > > > [FAILED] > > Waiting for zap to come online...Error: missing > > /dev/zap! > > > > If I do > > > > /sbin/modprobe zaptel > > > > I get: > > FATAL: Module zaptel not found. > > > > If I do > > > > /sbin/modprobe ztdummy > > > > I get: > > > > FATAL: Module ztdummy not found. > > FATAL: Error running install command for ztdummy > > > > Also, if i run: > > > > /etc/init.d/zaptel reload > > > > I get: > > > > Reloading ztcfg: Notice: Configuration file is > > /etc/zaptel.conf > > line 0: Unable to open master device > '/dev/zap/ctl' > > 1 error(s) detected > > > > > [FAILED] > > > > If I go back to /usr/src/zaptel-1.2.4 and I do > > > > make ztdummy > > > > I get: > > > > cc ztdummy.o -o ztdummy > > > /usr/lib/gcc/x86_64-redhat-linux/3.4.4/../../../../lib64/crt1.o(.text+0x21): > > In > > function `_start': > > : undefined reference to `main' > > ztdummy.o(.text+0xc): In function `ztdummy_timer': > > /usr/src/zaptel-1.2.4/ztdummy.c:154: undefined > > reference to `zt_receive' > > > ztdummy.o(.text+0x18):/usr/src/zaptel-1.2.4/ztdummy.c:155: > > undefined > > reference t > > o `zt_transmit' > > > ztdummy.o(.text+0x1f):/usr/src/zaptel-1.2.4/ztdummy.c:156: > > undefined > > reference t > > o `jiffies' > > ztdummy.o(.text+0x4d): In function `init_module': > > include/linux/slab.h:93: undefined reference to > > `malloc_sizes' > > ztdummy.o(.text+0x52):include/linux/slab.h:93: > > undefined reference to > > `kmem_cach > > e_alloc' > > ztdummy.o(.text+0x6a): In function `init_module': > > /usr/src/zaptel-1.2.4/ztdummy.c:232: undefined > > reference to `printk' > > > ztdummy.o(.text+0x197):/usr/src/zaptel-1.2.4/ztdummy.c:192: > > undefined > > reference > > to `zt_register' > > > ztdummy.o(.text+0x1a9):/usr/src/zaptel-1.2.4/ztdummy.c:239: > > undefined > > reference > > to `printk' > > > ztdummy.o(.text+0x1b5):/usr/src/zaptel-1.2.4/ztdummy.c:240: > > undefined > > reference > > to `kfree' > > > ztdummy.o(.text+0x1e2):/usr/src/zaptel-1.2.4/ztdummy.c:261: > > undefined > > reference > > to `jiffies' > > ztdummy.o(.text+0x23d): In function `init_module': > > include/linux/timer.h:87: undefined reference to > > `__mod_timer' > > ztdummy.o(.text+0x255): In function `init_module': > > /usr/src/zaptel-1.2.4/ztdummy.c:286: undefined > > reference to `printk' > > ztdummy.o(.text+0x27c): In function > > `cleanup_module': > > /usr/src/zaptel-1.2.4/ztdummy.c:298: undefined > > reference to `del_timer' > > > ztdummy.o(.text+0x288):/usr/src/zaptel-1.2.4/ztdummy.c:303: > > undefined > > reference > > to `zt_unregister' > > > ztdummy.o(.text+0x294):/usr/src/zaptel-1.2.4/ztdummy.c:304: > > undefined > > reference > > to `kfree' > > ztdummy.o(.text+0x39): In function > `ztdummy_timer': > > include/linux/timer.h:87: undefined reference to > > `__mod_timer' > > ztdummy.o(.text+0x2b0): In function > > `cleanup_module': > > /usr/src/zaptel-1.2.4/ztdummy.c:310: undefined > > reference to `printk' > > ztdummy.o(__param+0x10): undefined reference to > > `param_set_int' > > ztdummy.o(__param+0x18): undefined reference to > > `param_get_int' > > collect2: ld returned 1 exit status > > make: *** [ztdummy] Error 1 > > > > Any ideas? I know I posted things in some wrong > > order here, but when I > > actually did them as a part of the install > progress: > > I followed the order > > layed out in chapter 3 of the book. > > > > Thanks for any assistance. > > > > Take care, > > Sina > > > > ___ > > --Bandwidth and Colocation provided by > Ea
Re: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel
did you uncommnet # from before ztdummy ? --- Sina Bahram <[EMAIL PROTECTED]> wrote: > Hi all, > > I hope everyone is doing well. I just joined the > list, and I've really > enjoyed all I have read about asterisk so far. > Unfortunately, I'm having a > bit of trouble implementing this thing :). > > By the way ... I did my best to search the forums, > and also to use google > extensively, and while I have found pages with > people with the same problem, > ... The fix suggested on those sites, didn't work > for me. > > Here's what I have: > > Results of uname -r: > 2.6.9-22.0.2.106.unsupportedsmp > > Arch: > X86_64 > > If you need more specs on the machine or OS, please > let me know. > > I downloaded and have been following the asterisk > book, and in chapter three > I followed all the instructions on downloading the > sources, untarring them, > and so forth. > > Zaptel compiled without a hitch, as did the rest of > the asterisk packages. I > modified udev, and I restarted the box: ... I did: > > /etc/init.d/zaptel start > > I get: > > Loading zaptel framework: FATAL: Module zaptel not > found. > > [FAILED] > Waiting for zap to come online...Error: missing > /dev/zap! > > If I do > > /sbin/modprobe zaptel > > I get: > FATAL: Module zaptel not found. > > If I do > > /sbin/modprobe ztdummy > > I get: > > FATAL: Module ztdummy not found. > FATAL: Error running install command for ztdummy > > Also, if i run: > > /etc/init.d/zaptel reload > > I get: > > Reloading ztcfg: Notice: Configuration file is > /etc/zaptel.conf > line 0: Unable to open master device '/dev/zap/ctl' > 1 error(s) detected > > [FAILED] > > If I go back to /usr/src/zaptel-1.2.4 and I do > > make ztdummy > > I get: > > cc ztdummy.o -o ztdummy > /usr/lib/gcc/x86_64-redhat-linux/3.4.4/../../../../lib64/crt1.o(.text+0x21): > In > function `_start': > : undefined reference to `main' > ztdummy.o(.text+0xc): In function `ztdummy_timer': > /usr/src/zaptel-1.2.4/ztdummy.c:154: undefined > reference to `zt_receive' > ztdummy.o(.text+0x18):/usr/src/zaptel-1.2.4/ztdummy.c:155: > undefined > reference t > o `zt_transmit' > ztdummy.o(.text+0x1f):/usr/src/zaptel-1.2.4/ztdummy.c:156: > undefined > reference t > o `jiffies' > ztdummy.o(.text+0x4d): In function `init_module': > include/linux/slab.h:93: undefined reference to > `malloc_sizes' > ztdummy.o(.text+0x52):include/linux/slab.h:93: > undefined reference to > `kmem_cach > e_alloc' > ztdummy.o(.text+0x6a): In function `init_module': > /usr/src/zaptel-1.2.4/ztdummy.c:232: undefined > reference to `printk' > ztdummy.o(.text+0x197):/usr/src/zaptel-1.2.4/ztdummy.c:192: > undefined > reference > to `zt_register' > ztdummy.o(.text+0x1a9):/usr/src/zaptel-1.2.4/ztdummy.c:239: > undefined > reference > to `printk' > ztdummy.o(.text+0x1b5):/usr/src/zaptel-1.2.4/ztdummy.c:240: > undefined > reference > to `kfree' > ztdummy.o(.text+0x1e2):/usr/src/zaptel-1.2.4/ztdummy.c:261: > undefined > reference > to `jiffies' > ztdummy.o(.text+0x23d): In function `init_module': > include/linux/timer.h:87: undefined reference to > `__mod_timer' > ztdummy.o(.text+0x255): In function `init_module': > /usr/src/zaptel-1.2.4/ztdummy.c:286: undefined > reference to `printk' > ztdummy.o(.text+0x27c): In function > `cleanup_module': > /usr/src/zaptel-1.2.4/ztdummy.c:298: undefined > reference to `del_timer' > ztdummy.o(.text+0x288):/usr/src/zaptel-1.2.4/ztdummy.c:303: > undefined > reference > to `zt_unregister' > ztdummy.o(.text+0x294):/usr/src/zaptel-1.2.4/ztdummy.c:304: > undefined > reference > to `kfree' > ztdummy.o(.text+0x39): In function `ztdummy_timer': > include/linux/timer.h:87: undefined reference to > `__mod_timer' > ztdummy.o(.text+0x2b0): In function > `cleanup_module': > /usr/src/zaptel-1.2.4/ztdummy.c:310: undefined > reference to `printk' > ztdummy.o(__param+0x10): undefined reference to > `param_set_int' > ztdummy.o(__param+0x18): undefined reference to > `param_get_int' > collect2: ld returned 1 exit status > make: *** [ztdummy] Error 1 > > Any ideas? I know I posted things in some wrong > order here, but when I > actually did them as a part of the install progress: > I followed the order > layed out in chapter 3 of the book. > > Thanks for any assistance. > > Take care, > Sina > > ___ > --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 > __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://list
Re: [Asterisk-Users] D-Link DVG-1402S
Come on.! Don't tell me no one has ever had a problem on this model with asterisk? Live it up guys... and make a few comments Cheers Stephen Stephen Arulraj wrote: Anyone knows how to hook this up with Asterisk? ___ --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] Program Buttons on Cisco 79xx Phones
On Sun, 5 Mar 2006, Michiel van Baak wrote: On 02:08, Sun 05 Mar 06, [EMAIL PROTECTED] wrote: sccp and asterisk has some err.. real annoying bugs at the moment, where ciscos running SIP don't have these problems. Yeah, but still I can live with that because all the other things make up for that. The only annoying thing I have is the GroupPickup not working. Besides that they do all the SIP version does, and more. chan-sccp also does not handle reinvites yet. for large deployments this may be unacceptable. -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
RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe
Hi James, I am definitely interested in the card and also in the results of your testing. Regards, David LEXNET PTY LTD [e] [EMAIL PROTECTED] [m] 0411 172 667 Mail: PO Box R1180 Royal Exchange, Sydney NSW 1225 > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > James Harper > Sent: Saturday, 4 March 2006 12:03 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] MultiBRI in Australia - found one - maybe > > I may have found a source of an A-Ticked HFC 4BRI PCI adapter > in Australia, and will be testing one next week if all goes > well. I don't want to post the details of the reseller online > unless invited to do so, so if nobody replies and says they > are interested then I won't :) > > I'll follow up once I've tested it. > > Let me know if you want the details. > > James > ___ > --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 > > > -- > Internal Virus Database is out-of-date. > Checked by AVG Free Edition. > Version: 7.1.375 / Virus Database: 267.15.11/264 - Release > Date: 17/02/2006 > > -- Internal Virus Database is out-of-date. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 267.15.11/264 - Release Date: 17/02/2006 ___ --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] Can log into the mailbox from Soft-phone , but not from Hardware Phone
Thanks Alberto I am able to login now , I had used the option "dtmfmode=auto" thanks Joseph John http://www.voip-info.org/wiki-Asterisk+sip+dtmfmode --- Alberto Sagredo <[EMAIL PROTECTED]> wrote: > I suppose you are using 1.2.4 asterisk version > > Maybe is not sending dtmf tones as rfc2833 and > inband mode is not being > detected by your asterisk box. > > Im a wrong? Could you try to configure dtmf tones on > your softphone? > > John Joseph escribió: > > Hi > > I am using asterisk 1.4 on RHEL4 > > I am sending this mail to the mailing list , > to > > enquire wheter any one had faced simillar problem > > which I am facing now > > I am facing a problem which is not able to > solve > > or understand , the problem is that I cannot log > into > > the mailbox from a VoIP hardware phone , while I > am > > able to login to the mail box using soft-phone for > the > > same users > > Has anyone faced this kind of problems for > mail > > Can log into the mailbox from Soft-phone , but > not > > from Hardware Phone > >I am using hardware phone from grandstream > "Budge > > Tone > > -100 " > >and another D-Link phone DPF-140S > > > > Would like to get feed-back > >Thanks > >Joseph John > > > > > > > > > > > > ___ > > > To help you stay safe and secure online, we've > developed the all new Yahoo! Security Centre. > http://uk.security.yahoo.com > > ___ > > --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 > ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com ___ --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] Program Buttons on Cisco 79xx Phones
On 02:08, Sun 05 Mar 06, [EMAIL PROTECTED] wrote: > On Sun, 5 Mar 2006, Michiel van Baak wrote: > >On 20:52, Sat 04 Mar 06, [EMAIL PROTECTED] wrote: > >>We're still waiting for a SIP-enabled 7970... > >>The newer model phones (7941g/ge, 7961g) are sccp-only. Seems a step > >>backwards to me. > >why? > > If cisco really is moving towards SIP as claimed earlier, then releasing > new phones which are sccp-only is a step backwards from that goal. > > If cisco really is moving towards SIP as claimed earlier, then cisco > should release SIP images for 7970, as they did with 7960 and 7940. I agree that they should provide SIP. And indeed releasing sccp only while stating you are switching to SIP sounds conflicting. > > >I had my phones running on SIP, got chan-sccp and started > >experimenting with it. > >All my phones are running SCCP now. The phones respond > >faster, you have more options etc. > >Of course it would be nice if they offer SIP so people have > >a choice, but I really think the chan_sccp is the way to > >have these phones work. > > This only means sccp is currently better for cisco phones, it doesn't mean > sccp is a better protocol. Agreed. > > sccp and asterisk has some err.. real annoying bugs at the moment, where > ciscos running SIP don't have these problems. Yeah, but still I can live with that because all the other things make up for that. The only annoying thing I have is the GroupPickup not working. Besides that they do all the SIP version does, and more. > > Given a choice I'd run SIP, if only to have the phones able to talk to > each other and gateways if the PBX dies for any reason. sccp can't do > that -- if you lose the PBX you totally lose all functionality on all > your sccp phones. If you loose the PBX there are more problems. I don't know how the SIP image handles this, but I do know the SCCP image can be configured to failover to a second PBX. That way we can continue everything we do when 1 of our PBX boxes decides to die. As you can see, it all comes down to a matter of personal taste. That's why they should release SIP, just to give you the ability to choose. Coming back to the OP question, the SCCP image gives you a lot more control over the buttons. It involves some altering in a .c file, but there you can specify your layout as you want it. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called 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] incoming calls dropout on PRI over TE110p
Just trying to think - are you using the standard E1 setup from ATP? I have found that the settings on their website work pretty well. Also - have you tried to put an answer in your dialplan? That might keep the dialplan open.. later, PaulH - Original Message - From: "Paul C" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, March 01, 2006 5:15 PM Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p > > Paul C wrote: > >> I am running Asterisk 1.0.9 and have been running all my calls through a > >> VSP over a IAX2 trunk however we have recently purchased and connected a > >> TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make > >> outgoing calls via it fine, however incoming calls are dropped after a > >> few seconds ( or as soon as a command like Playback, or the call is > >> picked up if forwarded to a SIP extensions ). > > >> SNIP << > > > > > overlapdial should usually be no in my experience. > > > Okay I've turned that to no with no change. I've just got off the phone to > Optus and apparently they had a client in melbourne last week and they fixed > the problem by turning crc checking off at the optus end. I don't suppose > that was anybody on here ? > > ___ > --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] Can log into the mailbox from Soft-phone , but not from Hardware Phone
I suppose you are using 1.2.4 asterisk version Maybe is not sending dtmf tones as rfc2833 and inband mode is not being detected by your asterisk box. Im a wrong? Could you try to configure dtmf tones on your softphone? John Joseph escribió: Hi I am using asterisk 1.4 on RHEL4 I am sending this mail to the mailing list , to enquire wheter any one had faced simillar problem which I am facing now I am facing a problem which is not able to solve or understand , the problem is that I cannot log into the mailbox from a VoIP hardware phone , while I am able to login to the mail box using soft-phone for the same users Has anyone faced this kind of problems for mail “ Can log into the mailbox from Soft-phone , but not from Hardware Phone “ I am using hardware phone from grandstream "Budge Tone -100 " and another D-Link phone DPF-140S Would like to get feed-back Thanks Joseph John ___ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre. http://uk.security.yahoo.com ___ --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] 20 seconds til voice transmission starts
Hello everybody, I'm experiencing a strange problem with my Asterisk. I hope you can help: Asterisk is running at my company behind NAT. Ports 5060 and 1-2 are being forwarded to it. I have put the router's external IP-address into externip in sip.conf. At home I'm using an AVM FritzBox Fon WLAN 7050 which is registered with the Asterisk at my company. When I try to call Asterisk (or a phone connected to the attached legacy-pbx) from home, it's ringing normally and I can hear my opposite. But it takes about 20 seconds until my opposite hears me! When I call the same number again staight after, everything is working fine from the beginning. Also, calls from the company to my home are working perfectly. I'm greateful for any tips! Regards, Lius ___ --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] Can log into the mailbox from Soft-phone , but not from Hardware Phone
Hi I am using asterisk 1.4 on RHEL4 I am sending this mail to the mailing list , to enquire wheter any one had faced simillar problem which I am facing now I am facing a problem which is not able to solve or understand , the problem is that I cannot log into the mailbox from a VoIP hardware phone , while I am able to login to the mail box using soft-phone for the same users Has anyone faced this kind of problems for mail Can log into the mailbox from Soft-phone , but not from Hardware Phone I am using hardware phone from grandstream "Budge Tone -100 " and another D-Link phone DPF-140S Would like to get feed-back Thanks Joseph John ___ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre. http://uk.security.yahoo.com ___ --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] Program Buttons on Cisco 79xx Phones
On Sun, 5 Mar 2006, Michiel van Baak wrote: On 20:52, Sat 04 Mar 06, [EMAIL PROTECTED] wrote: We're still waiting for a SIP-enabled 7970... The newer model phones (7941g/ge, 7961g) are sccp-only. Seems a step backwards to me. why? If cisco really is moving towards SIP as claimed earlier, then releasing new phones which are sccp-only is a step backwards from that goal. If cisco really is moving towards SIP as claimed earlier, then cisco should release SIP images for 7970, as they did with 7960 and 7940. I had my phones running on SIP, got chan-sccp and started experimenting with it. All my phones are running SCCP now. The phones respond faster, you have more options etc. Of course it would be nice if they offer SIP so people have a choice, but I really think the chan_sccp is the way to have these phones work. This only means sccp is currently better for cisco phones, it doesn't mean sccp is a better protocol. sccp and asterisk has some err.. real annoying bugs at the moment, where ciscos running SIP don't have these problems. Given a choice I'd run SIP, if only to have the phones able to talk to each other and gateways if the PBX dies for any reason. sccp can't do that -- if you lose the PBX you totally lose all functionality on all your sccp phones. -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
Re: [Asterisk-Users] Preferred editor(s) dialplan coding?
On Sun, 5 Mar 2006, Michiel van Baak wrote: On 21:22, Sat 04 Mar 06, C F wrote: vi here vim :) Combined with the syntax file for asterisk. http://www.bemroses.net/images/curves.jpg -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
Re: [Asterisk-Users] Preferred editor(s) dialplan coding?
On Fri, Mar 03, 2006 at 03:06:02PM -0500, S McGowan wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Hey all, > > First of all, hello again! Been a while since I've posted to the > list, but I've been here lurking and watching ;-) > > Anyway, I wanted to pose a general question to the list to see > if it turns up new suggestions for everyone/me. > > What is your preferred editor when coding in the dialplan? This > is mainly aimed at those of you who write the larger, more > complex dialplans. > > I've been using UltraEdit, but would like to see if I can't find > a better one, especially one with the ability to add-on and make > it more Asterisk friendly. > > What I'm looking for: > Let's try to give an answer that is slightly better than simply "$FAVORITE_EDITOR" and variations, please. a vim user myself. I don't use most of what you descvribe below, however: > 1. Syntax Highlighting, and ease of updating that highlighting Update asterisk.vim > 2. Auto-updating lists (like sidebars) with: (this is a total > WISH list) > Variables > Contexts > a Command list? Not sure how to implement this in vim Maybe through some tweak of ctags? > 3. SVN and/or CVS integration > 4. Project ability There are standard macros for that. See http://vim.org/ . As I don't use it myself, I can't comment on them. > 5. Macros (Macros are s handy!) naturally. try :h map > 6. Autocompletion (and autocomplete edit ability) Depends on the type of auto completion. Both vim and emacs have a simple words-based autocompletion that runs out to be a very powerful tool (in vim. If you have something working with ctags, then it can probably complete better spanning multiple files. I'm not sure regarding context-sensitive completion. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Program Buttons on Cisco 79xx Phones
On 20:52, Sat 04 Mar 06, [EMAIL PROTECTED] wrote: > We're still waiting for a SIP-enabled 7970... > > The newer model phones (7941g/ge, 7961g) are sccp-only. Seems a step > backwards to me. why? I had my phones running on SIP, got chan-sccp and started experimenting with it. All my phones are running SCCP now. The phones respond faster, you have more options etc. Of course it would be nice if they offer SIP so people have a choice, but I really think the chan_sccp is the way to have these phones work. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called 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] seg fault when skinny phone answers
On 20:19, Sat 04 Mar 06, Ryan Laginski wrote: > Downgrade to 1.0.10. I was unable to get the 12sp+ to work reliably in > 1.2.0-1.2.4 and had the same problem. You could try the chan-sccp.org driver for skinny/sccp The 12SP+ is listed as supported device. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called 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] Preferred editor(s) dialplan coding?
On 21:22, Sat 04 Mar 06, C F wrote: > vi here vim :) Combined with the syntax file for asterisk. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called 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
