Re: [Asterisk-Users] Grandstream GXP-2000
On Fri, 28 Oct 2005, Erick Baum wrote: We have 50 of these phones in one location and a couple remote phones. The problem seems to be caused by the volume settings on the phone. We have noticed that the echo seems to be worse when the volume is very high on the phone (not using speakerphone). We're still testing, but that's what we've been able to come up with so far. Which end experience the echo? The phone with the loud volume, or the other end? If it is the remote end that experience echo then I would suspect acoustic coupling from the earpiece to the microphone inside the handset. If this is the case there are a few solutions: - lower the volume (duh!) - try connecting another handset with a known good decoupling of the mic/speaker - get grandstream to use the software echo canceller when using the handset as well as when on the speaker phone. Peter ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] PRI value
On Thu, 29 Sep 2005, Jens [iso-8859-15] Kübler wrote: Have I to use also prilocaldialplan ? Can be left unknown. Explains what you expect as the incoming number to look like This is incorrect. It sets the TON/NPI pair for ougoing calling number presentation, i.e. the format of the caller id you send to the pstn. Incoming numbers are always accompanied by a TON/NPI pair. If you want to you can have Asterisk prepend different prefixes based on which TON/NPI was presented to you from the pstn. See e.g. nationalprefix etc. All this and in much more detail has been covered in this mailing list several times already. Peter ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Routes IPSEc And Asterisk.
On Wed, 14 Sep 2005, Carlos Arnt wrote: Everything is perfect, but i have in point B now a C Network that comes over Router. Point B com see and interact with Point C , but point A can´t In number : Point A = 192.168.2.0/24 Point B = 192.168.1.0/24 Point C = 192.168.3.0/24 Over ipsec i can ping from A to B and from B to A. From B i can ping C and from C ping B. But from A i can´t see C . Because C is not a VPN over Ipsec, it is only connected in my B network with address (192.168.1.254) I insert a route : route add -net 192.168.3.0/24 gw 192.168.1.254 Then everyone from point B can see Point C. How make point A see C too ?? Either add static routes at A, B and C or you need to start using a routing protocol such as RIP, OSPF, BGP etc to propagate the routing information. For a three-node network without redundant links static routes are a lot easier to set up. Make sure packet routing is enabled on B. These days many operating systems default to denying routing until explicitly enabled. Peter ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channalized T1 and PRI with Asterisk
On Mon, 5 Sep 2005, Ben Brown wrote: Any Particular recommendations on PRI protocol? I can chose from 4ESS, 5ESS, and NI1 This is not a direct answer to your question since I am mostly familiar with EuroISDN. Most PSTN providers in America seem to charge extra for every single feature on a PRI. You may want to check the pricing on features such as DID, CallerID etc. You are in a better bargaining position before signing the contract than after. Peter ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channalized T1 and PRI with Asterisk
On Mon, 5 Sep 2005, Ben Brown wrote: So the only difference with PRI is caller ID? What I am trying to determine is if the PRI has enough advantages to give up the voice channel used by the D channel. For what I am doing, caller ID is not necessarily that important for my application. The PRI signalling is more robust than any of the alternatives (except SS7). Call setup is faster, you can get DID, caller id and much better error reporting from the pstn. I would recommend against CAS or analoge connectes whenever isdn is available. Can Asterisk choose the context based upon the CallerID with a PRI? Yes, this can be acclomplished in the dialplan. Peter ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000 Poor sound Quality
On Thu, 1 Sep 2005, Jesus Mogollon wrote: We have all those problems and then some... after a while, the phone starts degrading: The ringing becomes lower and lower and there is a lot of stuttering in the conversation. Also, if I stop/start asterisk, half of the phones reconnect while the rest don't. I was using the same firmare as you but had to roll back to 1.0.1.9 http://1.0.1.9 because of the degrading issue. We have some polycoms connecting to the same server and they have no problems whatsoever so we know it's a problem with the GXP. These phones are definately NOT ready for prime time. I would stay away from them. Play it safe and use Polycoms or, if too expensive, maybe Sipuras 841. These GXP-2000s are pure evil. In fairness the 1.0.1.9 firmware works very well for us. The speakerphone has an unusable microphone, but that is not an issue for us. Other than that we have not experienced any problems. Peter ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ICD Features
On Wed, 31 Aug 2005, Hadar Pedhazur wrote: My only real problem with my current setup is that because I use Call Files to contact the Agents, I have no direct way to cancel ringing phones when the call has been bridged to another channel. You can use the Manager interface with the Originate command to do that. I think you can get back a call handle with the FastOriginate variant. The handle can be used to call Hangup to cancel the call. Peter ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP-2000 presence
On Tue, 30 Aug 2005, Anton Krall wrote: Speaking of GS.. I know polycom phones can eb rebooted with some script using sip_notify. Can GS phones do this also? You can reset the phones by requesting the right page from their built in web server as long as you know the admin password. Peter ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ICD Features
On Tue, 30 Aug 2005, Hadar Pedhazur wrote: Following up on a thread that I started about Agents/Queue and acknowledging calls before bridging them... Greg Boehnlein said that he was putting his efforts into ICD. I downloaded and installed ICD, and I can get simple queue and agent stuff working fine, and see that this new design is much cleaner and more powerful. We use ICD to handle a complex queuing situation. This involved a new distribution algorithm and hooks to the Asterisk management interface. ICD suffers quite a bit from over engineering. The complexity comes mainly from the same analysis that makes Asterisk overly comples - signalling is handled in parallell thread. There are some locking issues and races we have not been able to track down. For our use we no longer see any hangs, but from time to time agents enter an inconsistent state. Both Asterisk and the ICD subsystem would have been easier to implement and debug had they chosen a single-threaded model for the control flow and left the threads to handle the payload (voice etc). Anyway, the real point of this post is to point out that I am marginally surprised that there is close to zero traffic on this list regarding ICD, and I don't know if that's because no one uses it, no one has any problems with it (including wanting to get the new stuff working), or I'm just on the wrong list (I am not currently subscribed to -dev, but would head over there if this is an active topic on that list). ICD has its own mailinglist at [EMAIL PROTECTED] There is close to zero traffic there as well. I think the authors read it though. Peter ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Zaptel Leading Zero Problem With TE110P
On Tue, 23 Aug 2005, Gulzar Hussain wrote: yeah i am using chan_zap and i have tried all combinations of pridialplan and nationalprefix etc. What does a pri intense debug span XX show? Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P problem
On Mon, 22 Aug 2005, Guy C. Guckenberger wrote: Im using a TE110P as a trunk to a Panasonic KD-500 everything works well.but Im having this problem where one of the channels becomes blocked with a partial phone number after about two days. So if the channel that becomes blocked is channels 23 no calls can get in. If the channel that gets blocked is 22 then one call can get in. The only way to clear this is to reboot the server. This is the line in the log that seems to indicate what is going on. Aug 17 17:24:10 VERBOSE[1057]: -- Extension '6' in context 'from-pstn' from '4093' does not exist. Rejecting call on channel 0/23, span 1 The correct ext should be 6xxx. So apparently the Panasonic passed just a 6 and Asterisk is complaining. If you can handle the logging volume you can turn on pri debug span 1 or even pri intense debug span 1 and log all isdn messages. The logs may give you some clues as to what goes wrong, and who stops responding. Alos do a show channels to get an overall status and a zap show channel xx on the hung channel. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ring more than two isdn phones simultaneously
On Sun, 21 Aug 2005, Arik Funke wrote: If this is a limitation of asterisk, where is it located? In the chan_zap module? In zaphfc? I.e. would it help if I switched to mISDN? It is inherent in the channel-based structure of Asterisk. An audio channel is the basic measure used by applications such as Dial etc. This is shared by all channels as far as I know. One can imagine a special version of chan_zap that decouples the Asterisk channel entities from the actual B-channels. It would always generate a new fictitious asterisk channel structure and only link it to a real B-channel once the signaling indicated that a B-channel was required. I would be interested in how the commercial SS7 implementation for Asterisk works. SS7 would normally allow the audio paths to change in mid-call to potentially follow an altogether different route. Peter Peter Svensson [EMAIL PROTECTED] wrote: On Sat, 20 Aug 2005, Nico Giefing wrote: how many connection do you have from your asterisk to the old pbx? i think on 1 ISDN connection its only possible to let 2 phones ring, because 1 ISDN 2 channels... This is a limitation in Asterisk, not ISDN. Asterisk reserves a B-channel for each destination at the time of the CONNECT message. In the isdn world it is common to not actually allocate a B-channel until it is needed to carry audio. This also prevents Asterisk from letting the upstream switch select the B-channel on outgoing calls to the pstn. Asterisk is written this way since it uses the audio channel as the fundamental unit, with the D-channel as carrier of signalling for the individual B-channels. Another way to view ISDN is to consider the D-channel the fundamental unit, which can carry several audio streams as a side effect of the signalling. The first viewpoint resembles the traditional view of telephony as individual circuits, the second resembles the ISDN/SS7 view of the world. Changing Asterisk to be more ISDN-like is quite a lot of work. Peter - Original Message - From: Arik Funke [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 20, 2005 7:44 PM Subject: [Asterisk-Users] Ring more than two isdn phones simultaneously I am using a HFC-S card in nt mode with zaphfc driver to connect an internal isdn bus. I would like to signal an incoming call on, let's say, 4 phones. Right now I use: Dial(Zap/g1/21Zap/g1/22Zap/g1/24Zap/g1/23Zap/g1/29,,t) where g1 are my two isdn channels provided by HFC-S card an the 21,22,etc my internal numbers. When the command is executed however, only the first two specified phones ring. Etc. with the first channel 21 ist called, with the second 22. How can I get asterisk to signal to all phones with just one isdn channel? I am trying to duplicate the setup I had with my old isdn pbx with did above trick just fine... Maybe somebody can help me configure asterisk appropriately? Cheers, Arik PS: I gave following a try but without success: Dial(Zap/g1/21-29,,t) Dial(Zap/g1/21+29,,t) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Zaptel Leading Zero Problem With TE110P
On Sat, 20 Aug 2005, Gulzar Hussain wrote: I am having another strnage problem :) When I dialout on any number from asterisk, it use to add a leading zero in dialed number for e.g I dial a number 5832876 and when I check the tracer's result of PSTN switch that shows me call request for 05832876 thats why I can dial NWD and ISD calls but unable to dial local numbers What channel do you use? For chan_zap you may want to look at the pridialplan, especially pridialplan=dynamic and the nationalprefix etc. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ring more than two isdn phones simultaneously
On Sat, 20 Aug 2005, Nico Giefing wrote: how many connection do you have from your asterisk to the old pbx? i think on 1 ISDN connection its only possible to let 2 phones ring, because 1 ISDN 2 channels... This is a limitation in Asterisk, not ISDN. Asterisk reserves a B-channel for each destination at the time of the CONNECT message. In the isdn world it is common to not actually allocate a B-channel until it is needed to carry audio. This also prevents Asterisk from letting the upstream switch select the B-channel on outgoing calls to the pstn. Asterisk is written this way since it uses the audio channel as the fundamental unit, with the D-channel as carrier of signalling for the individual B-channels. Another way to view ISDN is to consider the D-channel the fundamental unit, which can carry several audio streams as a side effect of the signalling. The first viewpoint resembles the traditional view of telephony as individual circuits, the second resembles the ISDN/SS7 view of the world. Changing Asterisk to be more ISDN-like is quite a lot of work. Peter - Original Message - From: Arik Funke [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, August 20, 2005 7:44 PM Subject: [Asterisk-Users] Ring more than two isdn phones simultaneously I am using a HFC-S card in nt mode with zaphfc driver to connect an internal isdn bus. I would like to signal an incoming call on, let's say, 4 phones. Right now I use: Dial(Zap/g1/21Zap/g1/22Zap/g1/24Zap/g1/23Zap/g1/29,,t) where g1 are my two isdn channels provided by HFC-S card an the 21,22,etc my internal numbers. When the command is executed however, only the first two specified phones ring. Etc. with the first channel 21 ist called, with the second 22. How can I get asterisk to signal to all phones with just one isdn channel? I am trying to duplicate the setup I had with my old isdn pbx with did above trick just fine... Maybe somebody can help me configure asterisk appropriately? Cheers, Arik PS: I gave following a try but without success: Dial(Zap/g1/21-29,,t) Dial(Zap/g1/21+29,,t) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: [EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Initiating a transfer from an analog handset?
On Sat, 13 Aug 2005, Jamin W. Collins wrote: Is there a way to initiate a transfer using an analog handset? For instance I'm looking for a way to do something like the following: External call comes in and is answered by user A. After talking to the caller they determine that the caller really needs to speak to user B. Is there any way for user A to initiate a transfer to user B, using only their analog handset? Now to make things possibly more complex, is the above still possible if the analog handset is connected to a Zhone Zplex channel bank? For zap-connected (i.e. tdm) handsets you can use the hook flash to transfer calls. Alternativly you can enable the pound transfer feature (see the documentation for the Dial application). I am not familiar with the Zhone Zplex channel bank. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo problem -- network related?
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote: The problem is not sound setup related. It present even if microphone is disconnected. To repeat the question from Matt Riddell: Does he have Stereo Mix selected as a recording source? We have found the most common cause of a strong echo to be that the sound card is set to record the outgoing earphone signal. If you post inline it is much easier to see what your answers were to different questions or if you have missed one. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo problem -- network related?
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote: You don't get 'echo' on the network, you'd only get true echo connecting to analogue PSTN lines so as Matt pointed out it will sound set-up/card related. Yes, this would be the logical conclusion, although it is hard to beleive given what I hear. It sound like I am talking to myself at a pretty good quality. Actually echo quality is much better than other party. This sounds exactly like you are recording the outgoing audio. The windows drivers for some sound cards does that by default. Go to the mixer, select the recording options and enable all controls so they are not hidden. Check which sources are used for recording. E.g. all the Dell desktops we purchased this year have audio drivers that by default record the outgoing audio. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] real-time priority , -p switch
On Thu, 11 Aug 2005, Joseph wrote: In this case could somebody explain to me why run asterisk with ''-p switch? According to asterisk man explanation for -p is as follow: If supported by the operating system (and executing as root), attempt to run with realtime priority for increased performance and responsiveness within the Asterisk process, at the expense of other programs running on the same machine. Since Linux is not RTOS, why some folks are using this -p switch? It has no effect on standard Linux box. Linux is not a hard realtime os with guaranteed timing. What the -p flag does is to request the realtime scheduler. This means a process wil no longer be subjected to the stanadrd unix scheduling but rather use a strict priority scheduling. The net result is that once a process using the realtime scheduler is ready to run the kernel wihh schedule it as soon as possible. It will only be preempted by realtime processes of the same or better priority. With the addition of the lowlatency patches the worst case latency for userspace applications is very low. The remaining difference between a hard RT os is the guarantees it can make. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
On Thu, 11 Aug 2005, Geoff Manning wrote: We are having line noise issues in our Asterisk to legacy PBX integration. All SIP calls originating from IP phones sound crystal clear. All calls that originate from the legacy PBX (Isoetec 228) and route through the Asterisk and out SIP have a lot of line noise. I believe I have it pinned down to these Blue Alarm errors that I can see on the legacy PBX side. zttool shows no alarm but when I view the T1 stats on the Isoetec I see numerous Blue Alarms. A blue alarm sounds really strange. That indicates that the remote end (asterisk) in this case does not want to play at all. On a T1 it is sent as a continous series of unframed 1:s. I am not sure if asterisk ever sends a blue alarm (Alarm Indication Signal). Receiving a blue alarm is indicative of a serious problem. There should not be any audio at that time, since the blue alarm is actually a long unframed signal. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
On Fri, 12 Aug 2005, Geoff Manning wrote: OK. So I changed it to: span=1,0,0,d4,ami And the Blue Alarms are still occurring but now in conjunction with Slip errors. I feel like I am on the right track though. Which side shows the slips? I am not that familiar with T1, Are you sure the signalling between the pbx and asterisk is set the same on both? Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to fix a Blue Alarm?? Line Noise?
On Fri, 12 Aug 2005, Bruce Ferrell wrote: Hardware, possible. Unlikely to be cabling. It's usually a timing setting. The blue alarm is really a very specific alarm condition normally. It cannot quite see how it can be generated accidentally. Something along the path from the TE110P transmitter to the decoder in the pbx generates a AIS condition on the line. Theoretically a repeater or converter withing the pbx could generate the AIS condition on the line. Another option is that the pbx uses the term blue alarm for something other than the normal AIS signal on a T1. Disturbances and frame slips would normally generate a local OOF condition, eventually triggering a local red alarm and sending of yellow alarm indication to the remote side. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE110P flashing red/green when PRI connected
On Tue, 9 Aug 2005, Fredrik Lithén wrote: Yes, I tried that but it sent me a bit offtrack as it reported blue which I assumed was a clocksync problem, or at least, that was the info I could find. As far as I can tell zttool/zaptel uses the term BLue Alarm for the E1 term AIS (Alarm Indication Signal) which is a layer 2 signal sent by someone in the receive path (not necesarrily the PSTN end itself) indicating that no valid incoming signal is present. A bitstream is present at the receiver, though it is unframed and invalid (i.e. the receiver is seeing a transmitter that does not quite know what to transmit). This is different from a red alarm where there is no bitstream at all. One common cause for the blue alarm is when the PSTN end shuts down an interface that has many errors. This si common practice and is probably what happend when there was a mismatch in the crc setting. As it turned out, my provider didn't have error correction enabled so after have endured painstaking task of getting hold of the right person to talk to :) the problem was swiftly fixed. An additional point: in the original post you mentioned that your zaptel line looked like span=1,0,0,ccs,hdb3,crc4 which should almost certainly be span=1,1,0,ccs,hdb3,crc4 since you want to synchronize your clock rate to that of the PSTN provider. With your old setting you will occasionally get a slip. Normally not a major problem, but it can wreck havoc with data transmissions (Unrestricted Digital) or in some cases with the D channel signalling. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P flashing red/green when PRI connected
On Tue, 9 Aug 2005, Andrew Kohlsmith wrote: On Tuesday 09 August 2005 04:32, Peter Svensson wrote: A bitstream is present at the receiver, though it is unframed and invalid (i.e. the receiver is seeing a transmitter that does not quite know what to transmit). This is different from a red alarm where there is no bitstream at all. I thought that red alarm was when it wasn't receiving a properly framed signal, and it sent an unframed all-1s pattern to the far end. Yellow alarm was when it was seeing an unframed all-1s pattern and was then trying to send a properly framed signal to the far side? I believe you are correct regarding the red alarm. Red alarm is declared when a frame loss has persisted for more than 2.5s. It is a local alarm. A framing error is a neccesary consequence of a LOS. :-) Yellow alarm (Remote Alarm Indication) is sent when a frame error condition exists in the receiver. On a T1 it is sent in bit 2 of every frame (for D4) or through a pattern in ESF. For an E1 two separate errors indications are collectivly known as yellow alarm, loss of framing (sets the A bit) or loss of multiframeing (sets the Y bit). Blue alarm (Alarm Indication Signal) is sent when the remote end does not want to communicate. It is sent as unframed 1. I seem to remember blue and yellow alarm being the same thing bu tit's 6am here and the mind is very much foggy. :-) Blue alarm - the other end is either administrativly down or there is a disconnect between various layers somewhere along the receive path. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN DID
On Tue, 9 Aug 2005, Eric Wieling aka ManxPower wrote: Panitaxx wrote: yes. overlapdial=yes. You want it to be no. What would the reasons to want overlapdial=no on a pstn pri be? Since the pri will happily signal once the number is complete there should not be any downside to allowing overlap dial. Are there pstn switches that do no like it? For reception of the number from pstn the overlapdial flag should not make any difference. The incoming called party number is either sent enbloc (in which case the overlapdial flag makes no difference) which is the common case or as overlap digits in information elements in which case overlapdial=yes is essential. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE110P flashing red/green when PRI connected ... continued
On Tue, 9 Aug 2005, Fredrik Lithén wrote: Perhaps everything isn't as spiffy as I thought When running zttool the card still reports as internally clocked Zaptel.conf: # Global data span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 Zttool still shows the card as internally clocked and you have a working connection at the same time? The only explaination I can come up with is that the changes to zaptel.conf have not taken effect. There have been a lot of talk about ztcfg requireing a full power cycle for come changes. It should not be possible for your card to be internally clocked given the above configuration. And as pointed out by Peter I do get a lot of D-channel warnings ... Aug 9 16:21:25 NOTICE[1350]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 This can be related to bit slips. Another possibility is missed interrupts. These can be checked in /proc/zaptel/1. And furthermore, now I've discovered that all channels seem to reboot from time to time Aug 9 16:15:18 VERBOSE[1350]: -- B-channel 0/1 successfully restarted on span 1 ... Could this be a HW problem with either the wiring, the PC or something else? Asterisk will reset unused channels once an hour by default. This is intended to keep channel state mismatches between the two endpoints from persisting indefinitly. The is harmless most of the time. Some equipment does not like these resets and there is some disagreement whether it is fully within the specification to send these automatically. The interval can be changed or the feature disabled with the resetinterval option in zapata.conf. We have a pbx that resets the whole E1 span in response to these requests, including active channels, so we have had to disable these resets. The resets will also be performed if the isdn signalling has been down on a span. If you get slips or irq misses bad enough this is a possibility. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
On Mon, 8 Aug 2005, Kib Eki wrote: Hi, we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our old PBX. So now we could migrate to the * server. But, there are two things we can't live with: 1. A call from the outside to the old PBX is missing a leading 0 before the number. Ex: caller has number 0123456 - * routes to old pbx - old pbx sees 123456 as caller number. See internationalprefix, nationalprefix etc in the file zapata.conf. 2. A call made from a SIP client to the outside lacks the extension in the number: Ex: PSTN number is 6789-0. The extension 234 is not added to the PSTN number like 6789-234 when dialing out over the PSTN. Are you refering to the dialed number or the outgoing caller id (calling number)? Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P flashing red/green when PRI connected
On Mon, 8 Aug 2005, Fredrik Lithén wrote: I'm having difficulties getting up my TE110P (running as a E1) when I connect it to the PRI. If I start the server with a loopback connector everything seems fine and the led is green but when I connect it to the PRI the flashing starts The command zttool may tell you what is wrong. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
On Mon, 8 Aug 2005, Eric Wieling aka ManxPower wrote: Peter Svensson wrote: See internationalprefix, nationalprefix etc in the file zapata.conf. Those options are only available in BRIStuff. They have been in HEAD for quite some time. The 1.0.x-releaes are note really usable in a lot of situations. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
On Mon, 8 Aug 2005, Kib Eki wrote: 2. A call made from a SIP client to the outside lacks the extension in the number: Ex: PSTN number is 6789-0. The extension 234 is not added to the PSTN number like 6789-234 when dialing out over the PSTN. Again, trivial dialplan stuff. Your sip.conf will have the callerid for each SIP client and you can append that information to the outgoing CID. That is set correctly and works between sip clients. it is only a problem when i try to dial out over zap/g1. Most likely you and your provider are not in agreement on how the calling party number should be encoded (number of digits and which Type Of Number / Numbering Plan). The TON/NPI is set with the prilocaldialplan option. Make sure you send the expected number of digits. You may have to do a SetCallerId() before the dial. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is this echo problem down to IP Phone hardware?
On Sat, 6 Aug 2005, Angus Comber wrote: I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. The Grandstreams are much maligned, but they actually do a better job in this area than most products. As said above, if you are using this in a large space the echo canceller in the phone may not cancel a long enough echo to be very effective. If it fails to kill the echo in a small room something is wrong. * The room is 15 foot by 22 foot. Not massive. When you say something is wrong, what should I be looking at? I will buy a Cisco 7940 as suggested previously to see if the handset does make a difference. Grandstream claims to be working on integrating code to cancel acoustic echo from the speaker phone. For the handset the echo is not canceled but rather minimized through good handset design with a very small acoustic coupling from the earpiece to the microphone. The major problem for the Grandstream GXP2000 is the acoustic coupling from the internal speaker to the internal microphone, not the room echo. This is a complicated problem since the speaker is much more powerful then the handset earpiece and at the same time the microphone needs to pick up a weaker signal. Fortunatly the coupling parameters for such internal coupling are more or less constant and with a short delay, making it easier to cancel. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very complicated dialplans?
On Sat, 6 Aug 2005, Robert Goodyear wrote: Can you educate us all on the appropriate circumstances in which to use 'r'? Some devices (voip phones, softphones) do not generate in band progress information when ringing. You will quickly find out if a particular end device requires the 'r' option or not. You almost never want it enabled on a trunk line, only for terminal devices. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 64K ISDN call not passing thru
On Wed, 3 Aug 2005, Tim Connolly wrote: I'm trying to pass a 65K DATA call in one channel on my Digium TE411P to another channel on a different span. Any idea what could keep this call from going through? -- Accepting call from '' to '5444' on channel 0/1, span 1 -- Executing Goto(Zap/1-1, sendto-definity|5444|1) in new stack -- Goto (sendto-definity,5444,1) -- Executing Dial(Zap/1-1, ZAP/g2/5444) in new stack -- Requested transfer capability: 0x08 - DIGITAL -- Called g2/5444 -- Zap/49-1 is proceeding passing it to Zap/1-1 -- Channel 0/1, span 3 got hangup request -- Hungup 'Zap/49-1' == No one is available to answer at this time (1:0/0/0) -- Channel 0/1, span 1 got hangup -- Hungup 'Zap/1-1' Do a pri intense debug span 3. This will give you a dump of the isdn traffic, hopefully including the reason for the hangup. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell Servers
On Wed, 3 Aug 2005, Sascha Ferley wrote: http://www.digium.com/index.php?menu=compatibility What servers does one recommend though using ? Our company hates using HP junk, dell used to be a good choice for most of our stuff. IBM is way overpriced. Anyone have any suggestions? If you need many servers at once IBM can give you a resonable price and very good engineering. Our main pbx is a Fujitsu 1u box with an unusually small footprint. We selected the Fujitsu since it fit in the telco rack in our server room. We have had no problems with it, though we only have pri lines connected to it. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED]/Grandstream Call Transfer
On Mon, 1 Aug 2005, Phoneguy wrote: There are 2 methods blind and announced here you go: Blind:Call someone, or receive a call. Hit 'Trnf' The screen displays TRANSFER TO? and you hear a dial tone. The other end can still hear you, so don't say anything nasty. Dial the number and hit 'Send', caller is transferred (blind) Announced: a.. Be on a call b.. Push a LINE button that isn't in use (this puts the call on hold) c.. Dial the extension you wish to transfer to d.. Speak e.. Push TRNF I think you need another step: f.. Press the line button of the first call Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] delay on pri dialling when asterisk is spliced between E1-Pri and legacy pbx
On Tue, 2 Aug 2005, Frank Sautter wrote: Maik Schmitt schrieb: one of our customers which wants a soft transfer between his old pbx to asterisk and sip. the setup is as follows: telco ---pri--- asterisk ---pri--- legacy pbx everything is fine exept that when dialling from the legacy pbx it takes about 3 seconds before the asterisk start to dial. where does this delay come from? has it to do with 'overlapdial=yes'? This is normal behaviour if you use '.' in your extensions.conf. Use '!' instead and Asterisk will start dialing immediately. when i change '.' to '!' then the overlap digits get lost. this means the longest number dialled on my telco line is as long as there are abigous matches in the dialplan. isn't there a way to start dialling after one received enough digits to decide which path to dial and then still transmit the remaining (overlapping) digits? If you have overlap=yes on both legs this should work. What you are seeing is consistent with having overlap=no on the outgoing call leg. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P Cable Pin Out
On Wed, 27 Jul 2005, Paul Dracevich wrote: I have just got a TE110P card, and I need the cable pin out. The TE110P cards use the standard T1/E1 modular pinout. See http://www.samhassan.com/isdn60.gif. 1 Receive from pstn (tip2) 2 Receive from pstn (ring2) 4 Transmit to pstn (ring1) 5 Transmit to pstn (tip1) Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] existing ISDN PBX - asterisk - 2xBRI for IVR and SIP
On Tue, 26 Jul 2005, Alex Ongena wrote: I'am new to * and googled/read a lot, but did not find (yet) a lot of info to do the above. Some months ago, I did find a 'story' from somebody having put * between his PRI and current PBX as IVR, but I can not find it back :-( We have an Asterisk pbx between the PSTN and our old pbx. Lots and lots of details to get right, but just take your time and work through them. * Thing through your dialplan in advance. * Which number goes where. * How do you dial from one system to another. (This depends to a large degree on your existing pbx) * Most user prefer the dialplan at each pbx to be prefix-free (see http://en.wikipedia.org/wiki/Prefix-free_code) i.e. that you dont have both 345 and 3456 as valid extensions at the same time. * Get the details on the isdn signalling * How many digits and which numbering plan are incoming/ougoing called numbers and callerid delivered/expected. * Which signalling format (since you use BRI I expect it to be EuroISDN). * Get the timing right. * You will almost certainly accept timing from the PSTN and provide it to the other pbx. * Multi-bri cards can take the timing from the PSTN and provide an identical clock to the other pbx. This preevents bit-slips and is a good thing. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Play Dialtone - get digits
On Wed, 20 Jul 2005, Ed Greenberg wrote: I'd like to write a snippet of dialtone that plays dialtone and collects a specific number of digits into a variable. Sort of like READ but with a generated dialtone. Naturally, I want the dialtone to stop playing after the first digit. I can't find this anywhere. Only thing I can think of is a no-password DISA. Is this the correct method? Is there a better one? DISA would proably work, though it may be a hassle since the call will be sent into the disa context. Another option is to use READ with a filecontaining a recording of the dialtone. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP2000 resetting all the time
On Wed, 20 Jul 2005 [EMAIL PROTECTED] wrote: I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly appreciated. Which firmware do you use on the Grandstream? Does the phone loose the ip connectivity (test wiht ping) or only the sip registration? We have not seen any problems with 1.0.1.9 on the GXP-2000. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI got event: HDLC Abort (6) on Primary D-channel of span 1
On Wed, 20 Jul 2005, Paul Belanger wrote: Any to back my clams that asterisk is fine, I'm using the TE405P, with a different telco in my second span and it operates fine!! What span is your clock source? A TE405P card can only operate in one clock domain at a time. I.e. the same clock will be used on all of them. If both providers are connected to the same TE405P card you will most likely have a problem with one of them unless they in turn have their clocks locked to a common source. Two differently clocked sources wil manifest themselves as the occasional bit slip. I am not sure if a bit slip can cause the problems on the D channel you are seeing. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Panasonic KX-TD500
On Mon, 18 Jul 2005, Guy C. Guckenberger wrote: Anyone have any luck with connecting Asterisk to the Panasonic KX-TD500. I have Asterisk connected via crossover to the TE110P. We are able to make internal calls into the Asterisk Box but the PBX vendor (I know nothing about the KX-TD500) tells us it is not possible route DID over the trunk. I find this hard to believe. Anyone have any luck with this? It depends on what T1/E1 card you have in the Panasonic, I think. It most certainly depends on what you are trying to accomplish. On the 500 it _is_ possible to set up a dialplan that will route some extensions out over a PRI link. That is one of the qualitative differences between the KX-TD1232/KX-TD816 and the KX-TD500. So, what kind of trunk card do you have in the Panasonic and what do you expect it to do? Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Panasonic PBX -to- Sirrix BRI: Numbers gettingechoed/duplicated
On Fri, 15 Jul 2005, David Wilson wrote: Thanks for your reply. Would srx show ccmsgs 1 help ? I am not familiar with the Sirrix line of BRI cards. However, someone else on the list may be, or you may be able to diagnose the problem yourself. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Panasonic PBX -to- Sirrix BRI: Numbers getting echoed/duplicated
On Thu, 14 Jul 2005, David Wilson wrote: I have a Panasonic PBX linked to a Sirrix Quad BRI card that is running in TE (ptp) mode in a Asterisk box - this then links through Internet to another Asterisk box via IAX2. When a user on the Panasonic PBX system dials the extension of my Sirrix Asterisk box, Asterisk answers and says Please dial the number of the person you are looking for. This is done with cmd Background. When this user enters an extension number to call the numbers that get picked up by Asterisk are repeated/echoed. For example, if a user enters 19 at the voice prompt, Asterisk picks it up as 1199 and tries to then dial 1199 out to the remote Asterisk server. One possible cause is that Asterisk receives the digits both as isdn indications (out of band) and as dtmf. Are you sure you have answered the line? On a bri link audio can be passed even without the line being answered. Before the line is answered Asterisk can receive overlap digits. While in overlap reception mode in band (dtmf) digits are ignored. Yuo may want to enable overlap digits nn the link to the Panasonic. I am not familiar with this particular BRI card. If it is not based on zaptel then the configuration will have to be made elsewhere. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Panasonic PBX -to- Sirrix BRI: Numbers gettingechoed/duplicated
On Thu, 14 Jul 2005, David Wilson wrote:
Yes, as far as I know ? In that context I have the following:
[pabx2ip]
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,ResponseTimeout(3)
exten = s,4,Background,enter-ext-of-person
exten = _X.,1,Dial,IAX2/pmb/${EXTEN}
exten = t,1,Hangup
exten = i,1,Goto(s,1)
Should be OK ?
Yes, it should be ok. At this point Asterisk should have sent an Answer
message to the pbx and it should have stopped sending digit indications
in the control channel.
I am not familiar with the Sirrix cards. Can you get the equivalent of a
Zaptel pri intense debug for the data from that card?
Peter
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Re: [Asterisk-Users] Swedish CallerID?
On Sun, 3 Jul 2005, Josef Seger wrote: I have one other Dect phone connected to Digiums Card(TDM400P), an Ericsson DT 260. The Ericsson phone only supports true swedish standard CallerID (DTMF signalling before the first ring), and CallerID does not work for this phone:( I have measured the outgoing signal from the TDM400P card and I have confirmed thet NO DTMS signals is sent out. Is it possible to show swedish callerid on ordinary analog phones connected to the Digium card? If yes, can somebody see the problem in my configuration files? See the bug at http://bugs.digium.com/view.php?id=3866. The original poster did not respond in time. Perhaps you can help debug the patch there? If so, ask one of the maintainers to reopen the bug report. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Horrible MeetMe performance
On Sun, 26 Jun 2005, qrss wrote: It seems that the voip clock is slightly faster than the hardware clock that zaptel is timing from. The extra samples/second must be being buffered. Of course, this buffering would add up over time until the point that a VOIP sample is played back several seconds out of phase. Seems that either the zaptel clock source must be brought to closer tolerance, or the extra data that is being buffered must be thrown away in order to stay in sync. Any thoughts? The VoIP clocks are not synchronized to the PSTN, to each other or to a common reference. There is bound to be a frequency difference between several devices. Over time this will lead to dropped packages. At present Asterisk MeetMe does not drop superflous samples, instead choosing to queue them. See bug http://bugs.digium.com/view.php?id=4252. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- * - Euroisdn Italy
On Thu, 23 Jun 2005, Robert Rozman wrote: I'm pulling my hair down and getting bold :-) . I have Asterisk between Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff Asterisk) I'm trying to do just plain transfer of call from pbx to ISDN through Asterisk... It seems like PBX hangsup, when call is progressing with no apparent reason. I'd kindly ask for any advice or some working example for this On isdn side I also have a problem. Asterisk quite often says that it cannot create ZAP channel, although partticular span is reported up and active. I've also tried to connect loop between NT and TE port and call doesn't get through I'd really appreciate if anyone has any advice on this problem, or any experience or working example for italian ISDN and particular Panasonic PBX. Look at the logs from a pri intense debug span X to see what causes the lines to be hung up. Make sure progress detection is disabled. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] so many FXS ports :)
On Thu, 23 Jun 2005, Andrew Latham wrote: On 6/23/05, Seamus Abshere [EMAIL PROTECTED] wrote: That's what I'm confused about: * two 4 port FXS cards * one 24 port FXS channel bank both, neither, and if both -- why do you need the dual digium cards? shouldn't your channel bank just take MGCP or SIP or something? The idea is that the channel bank breaks a T1/E1/J1 in to channels. There are newer channel type devices that are just gateways. I am sure there are benefits of doing it each way. By not having any VoIP links you reduce the latency considerably. This reuces the risk of perceiving a far end reflection as echo, instead keeping it as a normal side tone. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream 100 pricing question
On Wed, 22 Jun 2005, Pavel Jezek wrote: I had gxp-2000 for testing some days, but features are (in current firmware) _very_ limited! phone does not have missed, dialed numbers, phone book, speakerphone is useless... Some of these features are in the 1.0.1.9 version that was released last week. Missed and dialed numgers are available, although not in a very good interface (press the left and right arrows while off-hook). They do have separate memories per configured account. Grandstream clains thay will address the speakerphone problems in an upcoming release. I think they need a more advanced echo canceler since the speaker and microphone are acoustically strongly coupled. Also expected in the near term is suppor for Subscribe/Notify. phone have nice backlight display and in-line power :-) but if you like features, grandstream is not for you... On the other hand Grandstream seem to care about what their users want, at least for minor features. Everything we asked for was included in the current release. They seem serious in their attempt to break into the higher end market. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?
On Tue, 21 Jun 2005, Leandro Morgado wrote: Steve Underwood wrote: Robert Rozman wrote: I'm getting unreliable dtmf recognition (it works fine for 4-5 digits, errors (duplicates) on more), when transferred inband from gsm gateway to NT port of quadbri under bristuffed Asterisk. We get these quite often. If there is any line noise asterisk will interpret it as the end of a digit and then detect the same digit again. We are connected to the pstn via isdn. The problem is with calls where the dtmf tones are a bit unclean, i.e. too much energy is in the overtones. Clean dtmf tones seem to be much more resistant to line noise. Out other systems are more accepting of slightly off-spec dtmf tones. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Panasonic KX-TD1232
On Mon, 20 Jun 2005, Dan Morin wrote: Can you let me know what hardware you are using and how the two systems are configured to work together? Thanks in advance. We have an E1 PRI card in the KX-TD1232 and a TE405P in the Asterisk box. The Asterisk box sits between the pstn and the KX-TD1232. Can anyone confirm that dialing 8 + the Trunk Group number will select a CO line in that trunk? Thanks in advance. The exact digits to dial to request a specific trunk group can be changed, but it defaults to 8. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Panasonic KX-TD1232
On Sun, 19 Jun 2005, Dan Morin wrote: If anyone has any experience with a Panasonic KX-TD1232 phone system, I would really like to talk to you for a few minutes. I have asterisk connected to a Panasonic system via FXS - CO ports. I'm trying to get the Panasonic configured so that if someone dials a number (9) while Intercom is selected, it will select a line in the correct trunk group (Asterisk lines, rather than PSTN lines), then the user can finish dialing the asterisk extension. We did our integration using isdn lines between Asterisk and the Panasonic. There are all sorts of problems with using extensions as trunk lines. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Dell PowerEdge SC420 interrupt issue
On Fri, 17 Jun 2005, Paul Redstone wrote: We're using an SC420 and using BRI with a quadbri Junganns card, with IAX softphones and one hardphone. Working well except that we sometimes get dropped connections between IAX and the server with a max retries exceed message, which comes from the chan_iax driver code. The BRI side of things looks like it is fine. I had been thinking it might be a network issue but now wonder if it is an interrupt or other background process issue causing a timeout on the Dell - hence my post as it might be the same cause as yours. We're about to concentrate on this hypothesis. If it is then it could perhaps be due to: You could try running Asterisk with realtime privileges and see if it makes any difference. This will make the userland code preempt any other userland code. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Panasonic KX-TD1232
On Tue, 14 Jun 2005, Amund Nygaard wrote: We have around 50 phones in our company, and I am playing with the thought to gradually go over to using sip services and ip-phones internally. However at first I would liked the Asterisk just to sit between the phone line and the Panaosnic, so I can take out one lin/number at a time to use ip phones. I am new to Asterisk, and haven't done much configuring of the PBX either. So I also wonder how difficult such setup is. We use today 4 BRI lines that connects us to the telephone network, would I then need 2xTE410P to put the Asterisk between the Panasonic and the phone network? We use Asterisk in this exact way. You will either need two quad-bri cards in the asterisk box or 1 TE410P in the asterisk box and a Panasonic PRI card for the KX-TD1232. The TE410P is a quad PRI, not a quad BRI. We use PRI on all links. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP2000 and hint LED's
On Fri, 10 Jun 2005, Peter Svensson wrote: On Fri, 10 Jun 2005, James Bean wrote: Peter seems to be on the ball more then me about these phones as grandstream gave me the standard replies, Peter do you know for sure if grandstream have a timetable for the function led's cause I need to rollout about 50 phones and need 6-7 led's for display, which means a snom220+expansion, and gxp2000 seems perfect if it worked. I am certain that at least some documentation mentioned that the buttons will provide subscribe/notify in the future. I will ask our distributor to see what the official Grandstream position is. I received word from Grandstream today. The subscribe functionality is expected to make the next release. It is expected to ship in 1-2 months. No promises, but it is apparently high on their list of requested features. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP2000 and hint LED's
On Fri, 10 Jun 2005, The VoIP Connection wrote: Have you received an updated tftp config template as well? We asked for and received one with a 1.0.1.9 early beta version. That is the entire package as it was submitted to us from Grandstream. We requested and received the template separate from the firmware release. Without the template the phones can not be mass-deployed easily. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clicks in audio with TE100P PRI
On Thu, 9 Jun 2005, Andrew Kohlsmith wrote: I also check if I'm loosing interrupts and everything seems ok. Also I pull out the TDM400 from the box. This tells me it's got nothing to do with the TDM400 or lost interrupts. It could be that the user-land side (i.e. Asterisk as opposed to Zaptel) does not run often enough. A similar issue went away once we tuned on the real time scheduling for the Asterisk process. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP2000 and hint LED's
On Thu, 9 Jun 2005, Julian J. M. wrote: I've just checked the download page, and the latest firmware available is 1.0.1.8. Where did you find 1.0.1.9? This phone has some nasty bugs, one of them being that the other end HEARS you after you press the Transfer button and you hear a dialtone. It doesn't send any message to asterisk so that it can play music on hold to the caller. It is a pre-release version, not the actual 1.0.1.9. We received it to test a fix for a problem we observed in 1.0.1.8. So far we have not encountered any bugs with this pre-release. You may want to ask Grandstream support when it will be released. Within 24 hours of us reporting new bug we received a firmware which fixed the problem. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP2000 and hint LED's
On Thu, 9 Jun 2005, Michiel van Baak wrote: Did that pre-release version fix that bug where the other party can hear you when you pressed the transfer button ? That bug is not present in the testing version. Pressing the transfer button gives music on hold from the server to the other party. Does it also enable the leds next to the speeddial buttons like the snoms ? We have not tried, but I doubt it. That is probably going into the release after this one. Also being worked on is echo handling and increased microphone volume for the speaker phone. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP2000 and hint LED's
On Thu, 9 Jun 2005, The VoIP Connection wrote: This is supposed to be the final version: http://www.thevoipconnection.com/Downloads/GXP2000_1.0.1.9/GXP2000_Release_1 .0.1.9.zip From the changelog they seem to have corrected all bugs/misfeatures we reported during our testing of 1.0.1.9. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP2000 and hint LED's
On Thu, 9 Jun 2005, Michiel van Baak wrote: I really like the way the gxp2000 looks. I even prefer them above the snoms when it comes to looks. The bugs and lacking functions prevent me from rolling them out @ customers tho. The leds would be great, but the bug with the transfer button not muting the mic is critical. Is that one fixed ? We have not experienced that situation with the pre-release so I assume it is fixed. I will download and patch a phone in our internal setup tomorrow when I'm at the office again. Is it worth the time and effort to patch it ? Yes. :-) Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP2000 and hint LED's
On Fri, 10 Jun 2005, James Bean wrote: Unfortunately not, Grandstream didn't admit to me that they were going to program the LED's like the snom SUBCRIBE/NOTIFY, they told me the LED's were additional incoming line indicators, not LED's for the function keys to be programmed. Which is a little stupid, if they don't do the LED's like the snom then the phone is really no better then the BT102, just with a bigger LED and multiple sip account capability. If you want the 1.0.1.9 firmware pre-release goto www.atp.org.au and on the main page near the bottom it gives you a link. Peter seems to be on the ball more then me about these phones as grandstream gave me the standard replies, Peter do you know for sure if grandstream have a timetable for the function led's cause I need to rollout about 50 phones and need 6-7 led's for display, which means a snom220+expansion, and gxp2000 seems perfect if it worked. I am certain that at least some documentation mentioned that the buttons will provide subscribe/notify in the future. I will ask our distributor to see what the official Grandstream position is. The feature can not be that hard to program. I can not see why they would not implement it. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP2000 and hint LED's
On Thu, 9 Jun 2005, The VoIP Connection wrote: This is supposed to be the final version: http://www.thevoipconnection.com/Downloads/GXP2000_1.0.1.9/GXP2000_Release_1 .0.1.9.zip Have you received an updated tftp config template as well? We asked for and received one with a 1.0.1.9 early beta version. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] gxp-2000 tftp cfg
On Wed, 8 Jun 2005, David Phelan wrote: If you download the configuration tool which I couldn't get working on my systemthere is a cfg template in there for 1.0.1.8 Oh, then they have added it, or we missed it the first time around. We have it running. We had to tweak the paths in the file encode.sh a bit to match our setup. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP2000 and hint LED's
On Thu, 9 Jun 2005, James Bean wrote: Has anyone got the hint function working, and maybe with the GXP2000. I don't think the current firmware release for the GXP-2000 supports SUBSCRIBE/NOTIFY. That functionality is to be released at a later date. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gxp-2000 tftp cfg
On Tue, 7 Jun 2005, marek cervenka wrote: can you someone post tftp template for gxp-2000? like http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Windows/Grandstream_Configuration_File_Template_1.0.6.x.txt I think it will be released with the 1.0.1.9 firmware. You may be able to get it by asking their support for it. YMMW. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disa - how it returns on user not dialing any numbers ?
On Mon, 6 Jun 2005, Robert Rozman wrote: I'd like to use DISA properly for my case - I'd like to handle it right, if user when in DISA doesn't dial any number - how does Asterisk return from DISA cmd ? The file app_disa.c is hardwired to hang up the call if too many incorrect passwords are attempted and when no valid extensions has been entered before the digittimeout expires. To change it the block under the reorder: label in app_disa.c is probably the easiest. Instead of playing tones and all that it could set a channel variable based on the k variable which seem to be the main state variable. The dialplan could then handle the various exit cases. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
On Mon, 6 Jun 2005, Peter Nixon wrote: On Monday 30 May 2005 13:28, Matteo Brancaleoni wrote: and , what is more interesting, they've omitted any reference to digium resellers and specified only distributors :( Yes. Our reseller info was removed. And some of our customers have been sold to directly.. Not a nice way to do business :-( On the other hand, as an end customer I rarely see the need for the resellers, except as a cost-adding man in the middle. The distributors usually add value in that they clear customs and keep stock. To me, the supply chain ideally is manufacturer - distributor - customer or manufacturer - distributor - integrator - customer. Resellers have their place in mass market products. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pricing for DS3000P
On Sat, 4 Jun 2005, Tom Fanning wrote: What's so special about Digium cards that makes them this expensive? $4000 for a PCB is extortion IMO! I'd say low volume and high development and certification costs. A contributing factor is what the market is willing to pay. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 4 port BRI options ?
On Fri, 3 Jun 2005, Remco Barende wrote: I am thinking of another solution for fax. I have an * box with one PRI card and I'm thinking of adding a quad BRI card in the same box. A separate box running fasx server software will also be equipped with a BRI card for faxing (I cannot use spandsp for various reasons, one of them being that we must log / register all faxes). By keeping the path all digital I'm hoping to avoid trouble with echo (cancellation), interrupted faxes etc. The path is all digital but the clocks on the pri and the bri cards are not synchronized. Every now and again a frame will be dropped or duplicated to make up for the frequency difference. I guess ecm faxes will handle this if the frequency error is small enough. It would be sweet if the timing ports on the digium pri cards and the junghanns bri cards were/are compatible. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Meeting VS Call Confrence
On Thu, 2 Jun 2005, Mohamed A. Gombolaty wrote: I was trying to make call confrence available but all the asterisk documents use the meeting room concept, where those who wanna meet have to dial an extension corresponding to the meeting room, while call conference actually means that I am on exten 100 I can dial exten 200 and add it to confrence and again dial 333 and add it to the confrence and so on. Is there any way to make call confrencing available and not meeting room concepts? There is a patch to add call out from within a meetme conference. See bug number 3405 on http://bugs.digium.com/. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pridialplan prilocaldialplan
On Mon, 30 May 2005, Remco Barende wrote: What exactly is the meaning / function of the pridialplan prilocaldialplan? Both set the two fields Type Of Number (TON) and Numbering Plan (NPI) markers on an outgoing isdn call. These two tell a receiving isdn switch how to interpret the accompanying digits. The pridialplan option controls the TON+NPI associated with the Called Party information element. This is the recipient of the call. The prilocaldialplan option controls the TON+NPI associated with the Calling Party information element. This is the originator to be presented to the receiving user (think CallerId). I've been trying to find out what the different possibilities for these settings are but couldn't find a clear answer. The possible parameters I could find are are : local,unknown,dynamic,national,international and maybe there are more? unknown : set TON to unknown and NPI to unknown. This instructs the receiving switch to interpret the digits according to the standard used by the pstn in that country, leading zeroes etc included. E.g. 00461234567 for a call to Sweden. This is what one should normally use local: Almost never used unless requested by your pstn provider. national : Interpret the digits as a national number, i.e. with an area code at the beginning, but without any escape digits. I.e. no leading zero or similar for the area code. International dial is not possible. This is the default in Asterisk and almost always wrong. In some pstn networks in the USA this is actually interpreted like unknown above and not according to the specification. international : A fully formed E.164 phone number. 461234567 would be a call to Sweden. Usable. dynamic : Parse the dialed number and try to find a matching prefix in the settings internationalprefix, nationalprefix, localprefix, privateprefix, unknownprefix. If matched, set the TON/NPI to the matched setting and strip the prefix. Having set nationalprefix=0 allows you to call Dial with e.g. 0461234567 and have it sent as TON=national, digits=461234567. Setting the *prefix variables listed in dynamic above will _add_ the prefix on inbound calls. This can make the parsing of incoming calls easier. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GSX-2000 - dead :-(
On Fri, 27 May 2005, Mark Elkins wrote: I tried to do an HTTP update from the Grand Stream web site... You upgraded the firmware over the Internet? You are braver than I am. I would have used a local http server. Is there a magic re-incarnation routine ? (Power on whilst holding down some buttons?, Sprinkling chickens blood?) Have you tried the Grandstream support? I chose an HTTP upgrade over TFTP - as I read that there were potential issues with TFTP at this firmware level. Tftpd upgrades work well for us on that particular phone. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie here. Tips on setting up 100 phones w anted.
On Fri, 27 May 2005, Colin Anderson wrote: It will be about 100 phones at about 20 locations all within about 4 miles of each other. Perhaps a more pressing question might be how you are going to backhaul Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100 metres reliably, and using Ethernet repeaters every hundred metres or so isn't practical. You will need a fiber backbone or something like that. What is your plan to create an Ethernet network to tie these locations together? I suppose he could use 10Base5 (Thicknet). That gives you a whooping 500m per segment. ;-) Realistically there are lots of options - fibers, free space optics etc. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie here. Tips on setting up 100 phones wanted.
On Fri, 27 May 2005, Mike Clark wrote: brandt Milczewski wrote: I work for a ski area. I currently use a 3Com Superstack for in our office. And an old small town phone system for up at the mountain. The phone system is dying and I'm hoping to bring IP to replace the old phones. It will be about 100 phones at about 20 locations all within about 4 miles of each other. I'm looking for tips on the types of phones to look at. Cost is secondary only to reliabity. Any tips? We have over 100 Polycomm IP 300/500s installed, and they work great. The 300 will save you $50+ per phone over the 500, but you can't beat the 500 for a quality deskset with a good speakerphone. For that number of phones buying or getting loaner samples is probably the way to go. Resonable alternatives include Polycom, Snom or possibly even the GXP-2000, though the latter depends somewhat how much trust you place on future software upgrades. In these quantities you should be able to get a good price on any of these phones. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Grandstream phones.
On Wed, 25 May 2005, Shane Burrell wrote: Anyone with any comments on DSS buttons and general phone features? The BLF (Busy Light Field) part of the DSS buttons are not active in the latest firmware. The microphone part of the speaker phone needs some work, possibly just software (too low volume and too much echo). The headset jack in the back is a non-standard 3.5mm phono jack and not one of the usual 2.5mm phono or modular 4c4p jacks nearly all headsets use. You will either have to get an adapter made or connect the headset in line with the handset via a headset switch. Pity. The Called Numbers and Received Calls functions are awkward to use. I did not find a Missed Calls function. Other than these misfeatures the phone works well. Multiple lines, attended transfer etc. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Windows IAX Softphone
On Mon, 23 May 2005, Kanuri, Seshu (Company IT) wrote: FireFly is the best of the IAX softphones. Other softphones do not work as good as FireFly. DIAX has many bugs still. DIAX Softphone disconnects with Windows DLL errors everytime there is a problem in the call like Asterisk Channel Not available etc. The signalling in Firefly is not robust in the facce of dropped packets. The retransmissions are not handled correctly, leading to lost singalling and hence lost calls. All the problems went away when we started using iaxclient-based softphones such as iaxcomm. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red Alarm TE110P
On Tue, 24 May 2005, Remco Barende wrote: I'm trying to setup a Wildcard TE110P with a PRI in The Netherlands. I get a Red Alarm on the line. Is there any way of debugging this? I've tried some configs that should work but without success. Is there any way of telling if the cabling is correct or what else the problem could be? Have you configured the card to E1 and not T1? Red alarm is usually a loss of carrier (or close enough most of the time). I.e. you are not seeing incoming signals from the remote end. Check thatthe card is set to E1, check the cabling and check that the remote end is up. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Red Alarm TE110P
On Tue, 24 May 2005, Remco Barende wrote: On Tue, 24 May 2005, Huddleston, Robert wrote: OK, but being from Europe I haven't got a clue what an American SmartJack is for :) Would that mean that I would have to hook up the TE110P to the HDSL device? If so, what sort of cable would be needed for that? HDSL is not the same as a E1. Sometimes E1:s are tunneled over HDSL to extend the range without the need for midspan repeaters. Does the HDSL device have an E1 port? If so, connecting to it using a standard ethernet cable should work. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set CallerID in zapata.conf with QuadBri or other solution with parallel call signalling
On Sat, 21 May 2005, Companity wrote: The sip phones and the internal phones on the PBX see the number of the calling party correctly (e.g. 040-987654321). Cause we can´t set a callerid to the public phone network (to show the calling party number), we want to show an extension of our numbers on our isdn-bri (asigned by Carrier, e.g. our numbers are 12345-0 to 12345-99). If we use our [snip] Most (or some?) operators will actually allow you to set the outgoing calling party number provided you have signed a separate agreement. ETS 300 092-1 by default requires a strict checking of the calling number (paragraph 9.3). An alternate method available by special ararngement (paragraph 9.4) allows the sending of unscreened calling numbers. In this later case Annex B allows the network to add the network provided calling party number after the unscreen part. Similar rules are in effect in the USA. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Konftel
On Thu, 19 May 2005, Dean Collins wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Thursday, 19 May 2005 7:55 PM Another and perhaps easier option for wireless konference phones may be http://www.clearone.com/product_service/product_detail.php?prodid=127 and for larger rooms http://www.clearone.com/product_service/product_detail.php?prodid=198 Do you have a price? In Sweden they wireless ones are about $700. I guess for the flexibility you get it may be a good price, especially if you have a larger room so you occasionally need several devices working as one. Any ideas on quality? Have you used one personally? Not the wireless ones, but their wired conference phones work well for us. What is really nice about them is that several phones can either work standalone or be connected together to form a larger system for larger rooms when the need arises. This works as advertised. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ACD Methods
On Thu, 19 May 2005, Marshall, Ed wrote: Can anyone point me in the right direction of info regarding ACD methods available in Asterisk. As far as I can see there are time based ring strategies available but I cannot find any info regarding skills based routing or queue priorities. I don't think there is skill based routing in the standard acd queue mechanism. We have implemented a matching framework as an addon to the alternative queue implementation icd. We use it for a kind of skill based routing. A rework of the existing acd system seens to be in the works. At the moment the framework in app_acd makes it very hard to implement alternative strategies or a lot of other things one would want to add. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Expression in Extension
On Thu, 19 May 2005, Matthew Boehm wrote:
Hugh L. Johnson wrote:
Does ^ work as a NOT in an expression for extensions?
Are the following equivalent?
exten = _58[^389],1,dial(${${EXTEN}},${RINGLONG},tr)
exten = _58[0124567],1,dial(${${EXTEN}},${RINGLONG},tr)
Not sure which RegEx book you read but ^ (caret) stands for line beginning
not don't match.
Not inside a square bracket it doesn't. There a caret normally means
invert the set if it is in the first position. See e.g. man grep or
just about any other text describing standard regexps.
From reading pbx.c (EXTENSION_MATCH_CORE) it seems that asterisk does not
implement the inverted set inside []. That is ok, I guess, since the
regexps in Asterisk are among the most nonstandard ones anyone has ever
seen. :-)
Peter
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Re: [Asterisk-Users] Rack Mount Server Recommendations
On Thu, 19 May 2005, Michael B. Murdock wrote: Is there anywhere (or anyone) who has compiled some recommendations on rack mount servers for Asterisk? We are currently using Dell 2650 and Dell 2850 but are seeing some problems with the raid controllers failing and are now shopping for a suitable alternative. Ideally the server would be 19in rack mount, build with similar quality to the the Dell's, and have a -48VDC power supply option. Oh yeah, and run asterisk like a champ. The Fujitsu Primergy RX100 is nice and compact. I'm not sure about the -48V feed, ask Fujitsu. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Konftel
On Thu, 19 May 2005, Dean Collins wrote: Anyone seen these before? http://www.ascomnira.com.au/servlet/Display?p=100 wondering if there is a use with asterisk. Another and perhaps easier option for wireless konference phones may be http://www.clearone.com/product_service/product_detail.php?prodid=127 and for larger rooms http://www.clearone.com/product_service/product_detail.php?prodid=198 Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Spandsp: fax header
On Wed, 18 May 2005, Steve Underwood wrote: The header is always in the received image. The TIFF file contains exactly the same image that a receiving FAX machine would print out. I think he is refering to the remote fax id to be presented, not the header. I.e. the 20 digit user selectable number on the remote fax. The one often seen on the lcd of the receiving fax and so on. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Spandsp: fax header
On Wed, 18 May 2005, Steve Underwood wrote: Jean-Yves Avenard wrote: On my Brother's fax machine (MFC-8820D) today, I've received 3 faxes: all of them at the top showed the caller Fax identity. I received 2 faxes on Asterisk with spandsp, one from the same sender as earlier on the brother: there's nothing at the top. I wouldn't ask if it was obvious the data was inside the image, give me some credits for God's sake ! What you are describing is something I have never seen a FAX machine do. Typically, when somebody is sending a fax on the Brother unit, once the connection has been established the identity of the fax caller is then displayed on the Brother's LCD (and this has nothing to do with PSTN CallerID), what is displayed on the LCD will be printed at the top of each pages. This is this behavior I'm trying to reproduce with Asterisk/Spandsp. So you get the calling machine's number shown twice at the top of each page? Once in this extra header, and once in the normal header sent as part of the image? Weird. FAX machines don't normally do that. Does this extra header overlay a part of the page, or does it make the page one line longer? Both the fax machines at work have a setting to add a header on received faxes. The information it adds is the page number, the sending machine id, name (if the sending machine id is in the speed dial list), and date+time. The original poster is not alone in having machines work this way. Spandsp puts the calling machine's number in one of the tag fields in the TIFF headers. It puts several things in those tags - the name of the software which generated the file (spandsp), the hostname of the receiving computer, the far machine's ident, the far machine's maker and model (if they can be identified). Programs like tiffdump will show that information. Some image viewers also allow you to see it (don't ask me which ones off hand). Well, this should solve the original poster's problem. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream GXP-2000 and good support
We recently purchased a Grandstream GXP-2000 phone and I would like to share our experiences with it, especially out very good support experience. The phone was easy enough to set up. The phone was configured using a configuration file served via tftp. Creating the configuration file was a bit tricky since no template was released for this particular phone. Most options could be set from the template for the Budgetone/Handytone products. Most features work as advertised. In speaker phone mode the microphone volume is too low and the phone needs an integrated acoustic echo canceler. The speaker itself is nice and clear and work very well when just listening in. During testing we noted a problem with one-way audio when calls were placed almost back-to-back to the phone. We notified Grandstream through an email and though no more of it. After 3 hours we received a request for a tcpdump log, after 6 hours we received a confirmation that the support personnel had replicated the problem and within 24 hours we received a new firmware correcting the fix! I cannot emphasize enough the impression such quick and professional support makes. Especially since the problem had a workaround (we found that pressing hold twice cleared up the problem). No nonsense questions about whether we tried rebooting the phone. The new firmware was a beta of their next firmware, I guess. Some new features were added like: * multiple accounts now with user selectable names * auto-answer selectable per account * better display texts * even more configurable options We also received a configuration template that allowed complete control over the phone from the server. With the next firmware the phone does feel ready for deployment in a corporation. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream GXP-2000 and good support
On Wed, 18 May 2005, Anton Krall wrote: Peter.. I just bought a gxp 2000 and I wanted to know, how are you configuring them using templates? There is a template-binary config file compiler available from the download page at the Grandstream web site. Fill in the template and serve it via http or tftp. Some thing really dissapointed me and I sent some emails to GS about it. For example, ilbc is not supported eventhough it says so on the brochure... Seems they were eager to release the phone before finishing stuff... Like the conf button which does nothing. Yes, not all the features are enabled yet. However, at the moment the phone is a lot more capable than almost any sip phone costing twice as much. It works well enough for what we need it for, and if they add the promised features later on that is just iceing on the cake. Grandstream should have been more up front with what whas implemented and what was only planned. Also, you have such a good screen and callerid displays only half the info cause the other part is taken by Talking to or Calling... callerid could be 2 lines. Also, caller name is shown and not caller id num. Most of these issues were corrected in the testing firmware we received. Hopefully it will be released soon. You cant do 3 way conf calls. Not really a problem for us, we use meetme conferences exclusivly anyway. And the mad echo when using the spakerphone. Yes, as a speaker phone it is lacking. Still, not the most important feature by far. Speaker phones belong in a meeting room, not in an office, unless you have closed doors. Compared to Cisco phones the GXP-2000 wins hands down, if you factor in price at all. Given the large difference in price I'd say it beats most other phones as well. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phone Recommendations?
On Wed, 18 May 2005, John Mensel wrote: Hi all. I'm in the process of putting together a new Asterisk system as a proof-of-concept, and wanted to see which SIP phones all of you had the best luck using with Asterisk. I've just come off a very trying experience with some Cisco 7960s, and am looking for something else to round out the phones on our network. Try the Grandstream GXP-2000. With the upcoming firmware it fits our needs except for the receptionist. Note that we use headsets instead of speakerphones except in conference rooms. If a good two-way speakerphone is needed you should look at other phones. The price is hard to beat. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Run Script when originator hangs up the phone
On Wed, 18 May 2005, Erik Sundberg wrote: Wonder if there was away to run a script/marco when the person who originates the call hangs up. I have use the g option in the dial application to continue running applications in the dial plan, but that only works if the person who is called hangs up first.. Use the 'h' extension. That is run when the current channel (the caller normally) hangs up. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing spool file ignored
On Tue, 17 May 2005, tim panton wrote: The 'if possible' thing relates to filesystem design. Almost all of the native UNIX filesystems support mv as an atomic action - but only within the same filesystem. (Imagine you create the file on one physical disk then 'move' it onto a different disk - the kernel has no option but to copy the file). A Posix-compliant filesystem has to perform moves as an atomic operation. The removal of the old file name and the creation of the new file name will from the point of view of any process be simultaneous. Also, if the destination file name exists that name will refer either to the old file or the new, there is no window where neither file or an incomplete file is visible to a process. Moves are not defined across mount points. The kernel will never actually copy the file when move is calld. That is implemented in use space by e.g. mv in fileustils etc. So create your file in a temp directory on the _same_ file system as the destination, then do the move. If your filesystem is remote (samba or nfs) or non unix native (FAT) then it just won't work. Nfs is a posix filesystem - moves withing a device are guaranteed to be atomic. I think is is defined for smb as well, though there may be restrictions and the atomicity may not be guaranteed. Moves are defined and do work with the fat filesystem under linux. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
On Tue, 17 May 2005, Steve Underwood wrote: In most hardware the clock you use is not provided by a crystal. Rather the crystal provides a reference for a pll. The conversion factor between the crystan and the derived clock is usually tunable. Nope. Its always a crystal. Its either a pullable crystal in a VCXO, or its pulse-stuffed. It is required by the ITU specs to settle within 50ppm of the correct frequency when there is no signal driving its PLL, but many are out of spec. This is neither here nor there for the matter under discussion. I'm coming from the PLMN background and GPS background, and most of these receivers, cpus etc derive their clocks from a crystal via a pll that synthesizes the actual clock. Actually using the crystal oscilations as a clock is almost never done where precision is needed (read PLMN and GPS) since the raw crystals are way to dependant on temperature, even if they are cut to minimize their temperature coefficient. Whether the actual clock on the Digium cards is tunable enough I do not know. There are quite a few references to programming the clock in the source. Have you ever seen a framer where software can tune it? Its a hardware thing. Not having looked at the schematics for any E1/T1 interfaces, no. Like I said, in almost all places where a high precision clock is used a crystal is only used to feed a reference to a pll. The actual pll is then programmable. If the framer clock in Digium's boards are driven directly from the crystal then yes, no reprogramming is possible. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Background() problem (with queue(), etc.)
On Tue, 17 May 2005, Seb Auriol wrote: In fact, this is what I'm doing at the moment on the production system, but we've had a complaint because it doesn't start at the beginning for each caller. This is pretty important because the file starts with something like Thank you for calling X. We appreciate your patience during this brief period... There was some talk (possibly with patches) 6-12 months ago about modifying the queue application to allow it to play sound clips. Perhaps inspiration can be drawn from the icd project, where the announcements are listed together with instructions on how the list should be repeated. The list includes position and time announcements as well. Asterisk already has a syntax that is usable - the zonedata tone definitions. An announcements definition could be: announce=!welcome,15,!position,15,!msg2,30,position,30,msg3,30,position,30,msg4,30 This would play the welcome file, wait 15s, announce the position, wait 15s, play the msg2 file, wait 30s, play msg3, wait 30s, position, wait 30s play msg4, wait 30, repeast to the second position announcement. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Forwarding / Redirect with PRI
On Tue, 17 May 2005, Lenwood S. Sawyer, III wrote: I have a PRI from Bellsouth going to my asterisk box with a Digium Wildcard TE110P. I would like to be able to use call forwarding without having to use two channels. Is it possible to use call redirect with a PRI. Does the BRIstuff package help at all? What flavour of switch are you connected to? For Lucent 5ESS 2 B Channel Transfer is implemented in libpri. I think it will be used automatically if the conditions are suitable (no dtmf detection going on etc). BRIstuff provides ECT and CD which are used by EuroISDN. These are called explicitly from the dialplan, not automatically like the 2BCT above. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
On Mon, 16 May 2005, Michael Welter wrote: Where is the clock source that the T1/E1 board, with 0 for timing, uses to generate the tx data stream? Is there a PLL on each board? Or is some central source used? For example, I have one system with two separate T100P cards--one for a telco T-1 (#1) and the other for a channel bank (#2). For timing, #1 (telco) is set to 1 and #2 (channel bank) is set to 0. How does card #2 get its timing to generate its tx stream? Does card #1 interrupt the CPU based on the retrieved clock stream, and the CPU drive the other boards based on #1's interrupts? The #1 card will derive its clock from the received stream from the telco. The #2 card will run on an internal free running clock. The two cards are not synchronized at all. For the 4-port cards there is an unused connector marked timing. Perhaps Digium intends to update the fpga to allow cross-board timing distribution at a later date. It is possible, though complicated, to synchronize the 2Mbit clocks on two unrelated cards by measuring the accumulated phase shift (difference in interrupt rate) over time and compensating, thus implementing a PLL in software. Digium has not shown any intereset in such a solution. It is not clear if the internal hardware clock generator can be fine tuned enough to implement this. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
On Mon, 16 May 2005, Rich Adamson wrote: It doesn't make any difference. The pcm data that arrives from the telco is buffered in the zaptel and/or asterisk code, and sent out the second T1 card as soon as it can. That buffering reduces (or eliminates) the need to sync one T1 card to another. However, if the clock on the second card were way off frequency, there could be a missed pcm frame from time to time. The missed frame would not even be noticed by users in Any frequency error will eventually lead to dropped frames, and this is as it should be. For digital transfers this can cause problems. For voice it is normally a non-issue. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
On Mon, 16 May 2005, Steve Underwood wrote: It is possible, though complicated, to synchronize the 2Mbit clocks on two unrelated cards by measuring the accumulated phase shift (difference in interrupt rate) over time and compensating, thus implementing a PLL in software. Digium has not shown any intereset in such a solution. It is not clear if the internal hardware clock generator can be fine tuned enough to implement this. How can that work? You can measure the error, but you have no ability to tweak the clock from software. Two cards could only be synced by hardware. In most hardware the clock you use is not provided by a crystal. Rather the crystal provides a reference for a pll. The conversion factor between the crystan and the derived clock is usually tunable. Whether the actual clock on the Digium cards is tunable enough I do not know. There are quite a few references to programming the clock in the source. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDMoE emulates a T-1= Is there a product tosimulate a PRI trunk? (Robert Goodyear)
On Fri, 13 May 2005, jltaylor wrote: Does the TDMoE only allow one T1 per segment? You can add an index to have several TDMoE links and thus several virtual T1/E1 links between two computers. TMDoE is mostly used to provide an interconnect with a low latency over ethernet. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound card Line-In as MOH source
On Thu, 12 May 2005, Chris Coulthurst wrote: Does someone have a link to step-by-step instructions to making the Line-In on the console sound card a MOH source? You can probably use the Remote MoH patch from http://bugs.digium.com/view.php?id=3565 Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
