[asterisk-users] Skewed RTP timestamps in SIP calls on Asterisk 1.4.18
Last week I migrated some of our servers to Asterisk 1.4.18 and we started seeing audio drops of several seconds during SIP calls. After investigating it we noticed that Asterisk was increasing the RTP timestamps abnormally during a conversation. I'm including a text file with a subset of the data collected by Wireshark that shows the problem (I have the complete packet capture if anybody needs it to analyze it). The Asterisk server is the one whose IP address ends in .38. If you look at the packet with the number 14910 (seq 23369) you'll see that the next packet from Asterisk (14919, seq 23370) increases the RTP timestamp from 77120 to 2280582632. We've tried enabling and disabling internal timing and the jitter buffer, but it made no difference whatsoever. I also added the patch present in ticket #10355 to Asterisk 1.4.18, but it didn't help. Has anybody else experienced a problem like this one? 14898 52.678422 63.215.27.5566.150.122.38 RTP PT=ITU-T G.711 PCMU, SSRC=0x352D1216, Seq=22216, Time=1043051951 14899 52.678576 66.150.122.38 63.215.27.55RTP PT=ITU-T G.711 PCMU, SSRC=0x404D77E4, Seq=23368, Time=76960 14909 52.698326 63.215.27.5566.150.122.38 RTP PT=ITU-T G.711 PCMU, SSRC=0x352D1216, Seq=22217, Time=1043052111 14910 52.699321 66.150.122.38 63.215.27.55RTP PT=ITU-T G.711 PCMU, SSRC=0x404D77E4, Seq=23369, Time=77120 14917 52.718417 63.215.27.5566.150.122.38 RTP PT=ITU-T G.711 PCMU, SSRC=0x352D1216, Seq=22218, Time=1043052271 14919 52.720938 66.150.122.38 63.215.27.55RTP PT=ITU-T G.711 PCMU, SSRC=0x404D77E4, Seq=23370, Time=2280582632 14921 52.721029 66.150.122.38 63.215.27.55RTP PT=ITU-T G.711 PCMU, SSRC=0x404D77E4, Seq=23371, Time=2280582792 14922 52.721052 66.150.122.38 63.215.27.55RTP PT=ITU-T G.711 PCMU, SSRC=0x404D77E4, Seq=23372, Time=2280582952 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some queries on g729 license.
if you're only doing IP takeover and have bound the licenses to each server separately. If you're sharing the storage, then that could pose a problem. Leo DatVoiz Singapore Pte Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and g729 licenses
MixMonitor must convert the G.729 streams to 16-bit Linear PCM (slin in Asterisk) in order to mix the inbound and outbound streams for a call. If you want to avoid this, you could try using the Monitor application which saves each stream as a separate file with the codec that was used for the call. You will have to mix the files externally though to listen to the complete call. On Wed August 30 2006 02:43, jurgen wrote: Hi, I recently bought a handful of g729 licenses and moved all my equipment over to use it. We terminate most of our calls with a provider that supports g729, so it's g729 all the way through from the phone on the desk to the provider. Asterisk works very well in passthrough mode, simply moving the bits from the phone to the provider. Good work. The problem happens when I record a call using MixMonitor. Even though it's recording natively in g729, a single call uses 2 decoders and one encoder! The only explanation I can think of for that is that MixMonitor is transcoding the g729 streams to something else, muxing them, then encoding the muxed stream out to g729. This seems ridiculous - why go through all that work and licenses? Does anyone know for sure what's going on here? I could go back to using Monitor, I suppose, but MixMonitor is somewhat less hacky. Thanks jurgen -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does anyone use T.38?
We have successfully used Sipura 2100 ATAs for this with an external fax machine connected to its FXS port. The Sipura is connected to a Cisco fax gateway right now, we haven't been able to test it with Asterisk yet. On Fri August 25 2006 06:58, Ricardo Carvalho wrote: Does anyone use T.38 for fax? If you use it, what hardware / software do you use? Thanks, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Gateway
We're using a Cisco 3660 as a T.38 gateway for fax reception, but as Asterisk does not support T.38 in pass through mode yet what we're doing is sending a SIP REFER message (via the Transfer application) to our SIP provider (when we detect fax tones) to redirect the call to the Cisco gateway. Carlos Alperin wrote: Did someone use a 26xx, 36xx or 53xx as a T38 Gateway? I need to know if we can register an ata like Sipura 2100 to the cisco equipment, or we need to register the cisco on asterisk in order to complete the circuit. The documentation on the Cisco is always referred to the call manager. All that I want is to send PRI to ATA and ATA to PRI T.38. Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Gateway
I have to check about the configuration as I'm not the one who did it. Bear in mind that what we did was for fax reception, so the SIP REFER is being sent to our fax provider, not to the Cisco gateway. The gateway just receives the call after it's been redirected. Carlos Alperin wrote: Is any way to get an configuration example of that? I know that we cannot send the T.38 through the Asterisk, that is I'm trying to avoid that, but there is no way to register the Sipura 2100 on the Cisco. It's not a Gatekeeper or a VoIP Server. However there is 1.2.7.1 patch for the T.38 passthrough, I didn't tried yet. How do you send that SIP REFER to the Cisco? Gracias, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose Comellas Sent: Thursday, June 15, 2006 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco Gateway We're using a Cisco 3660 as a T.38 gateway for fax reception, but as Asterisk does not support T.38 in pass through mode yet what we're doing is sending a SIP REFER message (via the Transfer application) to our SIP provider (when we detect fax tones) to redirect the call to the Cisco gateway. Carlos Alperin wrote: Did someone use a 26xx, 36xx or 53xx as a T38 Gateway? I need to know if we can register an ata like Sipura 2100 to the cisco equipment, or we need to register the cisco on asterisk in order to complete the circuit. The documentation on the Cisco is always referred to the call manager. All that I want is to send PRI to ATA and ATA to PRI T.38. Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk spanDSP / Faxing problem
) on Zap/1-1 Mar 25 13:54:18 DEBUG[17501] chan_zap.c: Updated conferencing on 1, with 0 conference users Mar 25 13:54:18 VERBOSE[17501] logger.c: -- Hungup 'Zap/1-1' I have received a fax from a different machine with this config, but I would like to be able to have the faxes pulled out from incoming calls and voice going to a digital receptionist . Thanx Thys de Wet Cape Town South Africa ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ChanSpy via external application
This problem has been already corrected in Asterisk 1.2. See this bug: http://bugs.digium.com/view.php?id=6009 On Monday 09 January 2006 00:51, Peter Fern wrote: Just implemented a similar feature here - apparently the chanprefix won't accept a full channel identifier, so I ended up dropping the last character (this works for me since all the sip delivery we want to monitor is to individual handsets - I won't be monitoring any channels that are delivered in bulk). I should really file this as a bug - I would think more useful behaviour in most cases would be to take a channel identifier rather than a prefix. Dov Bigio wrote: Hello, It didn't work... I used Data: SIP/dov.bigio-9949 which was the channel being used, and the call I received just had beeps... no conversation. According to the documentation on (http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy), ChanSpy doesn't take a channel as parameter, does it? Thank you very much!! Dov - Original Message - *From:* Giovanni Miano mailto:[EMAIL PROTECTED] *To:* Dov Bigio mailto:[EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Thursday, January 05, 2006 7:01 PM *Subject:* Re: [Asterisk-Users] ChanSpy via external application Use channel of your agent Channel: SIP/dov.bigio MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: 01.telecom Application: ChanSpy Data: SIP/234-ssnf Priority: 1 Cheers, Giovanni Miano 2006/1/5, Dov Bigio [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi, I have developped an application that monitors the status of my queues through the events triggered on the Manager Interface. This way, I can know the status of my Agent real time. Now, I have a new requirement that I must allow a manager to click on the Agent he wants to monitor and be able to monitor the call. My idea was to, when the user clicks on the Agent, I would Originate a call between his extension and the extension I have for ChanSpy, passing as parameter the Agent number. For testing this, I tried a call file on /var/spool/asterisk/outgoing Channel: SIP/dov.bigio --- This is me MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: 01.telecom Application: ChanSpy Data: Agent/5450 - This is the Agent I want to monitor Priority: 1 The problem is that ChanSpy doesn't accept Agent/ as parameter, just Agent. Is there a way to ChanSpy a specific know Agent? (Or at least to send via dtmf the Agent Number I want to monitor right after the ChanSpy application is called? Thank you very much! Dov ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP/VPN providers in Switzerland
Does anybody know of any VoIP provider in Switzerland (or other Euro country not far from it) that could give me a DID with VPN termination. What I need is to have a SIP or IAX connection encrypted inside a VPN (ipsec preferably) to make and receive calls. Fax support would be a huge plus. Thanks. -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using RxFAX and TxFAX together
I don't understand exactly what you're telling me, but I'm currently using TxFAX with an already generated TIFF file to send a fax to another machine that uses RxFAX to receive it. RxFAX and TxFAX are not running on the same machine. The only strange thing is that I'm using a SIP connection using the G.711u (ulaw) codec between both machines inside a LAN. On Tue November 15 2005 23:19, George Vagenas wrote: Juan Jose Comellas wrote: Has anybody ever used the TxFAX application to send a fax to RxFAX on another Asterisk installation. I'm trying to do just that and both apps remain blocked in the ast_waitfor_nandfds() function without transmitting anything. I am calling TxFAX with the 'caller' parameter. What is strange is that both apps block on a call to ast_waitfor() with a inifinite timeout. I've seen this in several other places in Asterisk and these calls are normally the source of hung channels. Is this correct? As far as i know connecting rxfax to txfax, doesn't work. Why don't you save first and then send? ___ --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 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using RxFAX and TxFAX together
Has anybody ever used the TxFAX application to send a fax to RxFAX on another Asterisk installation. I'm trying to do just that and both apps remain blocked in the ast_waitfor_nandfds() function without transmitting anything. I am calling TxFAX with the 'caller' parameter. What is strange is that both apps block on a call to ast_waitfor() with a inifinite timeout. I've seen this in several other places in Asterisk and these calls are normally the source of hung channels. Is this correct? -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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] Trying to clarify ideas about spands, libtiff FC4
I've been using spandsp 0.0.2pre18 and 0.0.2pre21 with libtiff 3.7.3, Asterisk 1.0.7 (now 1.0.9) and the Linux kernel 2.6.11.5 kernel with results that are good enough for me (I'm using fax over IP with the G.711 codec). On Sunday 23 October 2005 13:23, Carlos Alperin wrote: I spent more than 3 weeks, with some little help of people that belongs to this forum, and after try differents combinations of versions this is my conclusion: I tried RH9, FC4 FC4 64 I tried with CVS 1.0.2, and Stable 1.0.9 I tried with spandsp 0.0.2pre18, 0.0.2pre20 0.0.2pre21 Libtiff 3.5.7 libtiff devel 3.5.7 Libtiff 3.7.1 libtiff devel 3.7.3 (I couldn't find 3.7.1) My conclusion is: If I need to be able to use fax with Spandsp, app_rxfax.c app_txfax.c with libtiff 3.5.7 (and libtiff devel 3.5.7) there is no way to do that on FC4 (get conflict with GTK2+) So it looks like I have to go back to RH9 and at least upgrade to kernel 2.4.31, and try again. This is under the presumption that Spandsp, the rest are going to work. (Looking at the forum, that is not a 100% fact). It should be a way to save us a lot of time, if somebody can unify all the requeriments on each OS, so we can decide before to start which direction to follow. The reason for RH9 FC4 is because they're more familiar. But if someone can show me a working configuration, I don't hesitate to move the platform. By the way, the 64 bits platform still looks to be very unstable and not so fast to implement with Asterisk. To the digium support: I understand that your recommendation is to go to 2.6 kernel, but if I need to run spandsp, how to do that without libtiff 3.5.7. The general experience is libtiff 3.7.1 locks the asterisk when the machine boots. Please feel free to send every kind of disappointments opinions. That is going to feel me much better that no answers. (Even if you can show me how stupid I was doing all kind of mistakes) Regards, Carlos Alperin -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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 and SPA-841
Check the the phones and see if they have the 3.1.3a version of the firmware. If so, upgrade the firmware to 3.1.4a. I had a similar problem and after this they started working correctly. On Thursday 06 October 2005 12:10, Chris wrote: I have a SPA-841 and sometimes the audio is one way.I can hear the other person but they can't hear me. Has anyone had this before? Regards, Chris -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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 as H323 gateway
I have setup Asterisk with chan_oh323 to connect to the PSTN over H.323 in Buenos Aires, Argentina. Currently I'm using direct connections to the telephone company's (iplan) H.323 gateway, but I'm working on using an intermediate H.323 gatekeeper to take advantage of the telephone company's redundant servers. I think the telco uses Cisco hardware, but I'm not completely sure. We've just started using this, but it seems stable so far. On Tue October 4 2005 06:28, [EMAIL PROTECTED] wrote: Is there anyone who is currently using Asterisk as a production H323 gateway? And using which combination of asterisk and H323 (chan_h323, chan_oh323?) The main issue is interoperability with other H323 parties (Cisco AS53xx, Nextone, etc). Searching the mailing list it seems that both h323 and oh323 are not so stable, is it only an impression or using h323 is really not so advisable? Francesco Pellegrini [EMAIL PROTECTED] ___ --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 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No ringback tone generated by Asterisk with OH323 connections
I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in Asterisk? The relevant settings in oh323.conf are: [general] listenAddress=0.0.0.0 listenPort=1720 tcpStart=20001 tcpEnd=3 udpStart=20001 udpEnd=3 fastStart=yes h245Tunnelling=yes h245inSetup=yes inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 bandwidthLimit=2000 gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 The package versions I'm using are: asterisk1.0.9.dfsg-5 asterisk-oh323 0.6.6pre3-4 libopenh323-1.15.3c21.15.3-4 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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] No ringback tone generated by Asterisk with OH323connections
No, I wasn't. I can't believe I made that stupid mistake. It started working after I added the call to Answer(). Thanks for your help. On Friday 30 September 2005 11:53, Brian C. Fertig wrote: are you giving answer()? ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in Asterisk? The relevant settings in oh323.conf are: [general] listenAddress=0.0.0.0 listenPort=1720 tcpStart=20001 tcpEnd=3 udpStart=20001 udpEnd=3 fastStart=yes h245Tunnelling=yes h245inSetup=yes inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 bandwidthLimit=2000 gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 The package versions I'm using are: asterisk 1.0.9.dfsg-5 asterisk-oh3230.6.6pre3-4 libopenh323-1.15.3c2 1.15.3-4 -- Juan Jose Comellas ([EMAIL PROTECTED]) -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recommended wireless router to run Asterisk on OpenWRT
I want to start experimenting with Asterisk running on a router with embedded Linux using the OpenWRT firmware. Has anybody tried routers other than the Linksys WRT54G or WRT54GS for this purpose? What do you recommend? -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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] sipuras 841 bad sound
Have you tried upgrading the firmware? I had several problems with the outbound volume of these phones until I upgraded them. On Tuesday 20 September 2005 20:46, Anton Krall wrote: Hi Guys! I have a problems with some sipuras 841 and asterisk 1.0.9. Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with steve's unicall. Everything compiled fine and in fact I can make and receive calls but I have a problem with bad sound when the sipuras call the outside E1's lines. I can listen to the caller without problems but they heard me with a choppy sound as if you were losing frames or cutting off. Calls between internal sipuras sound good (eventhough the speaker and headset sound comes and goes, for example, when you start talking, seems as if the sipuras takes a few seconds to catch up with you on volume so the remote user listen to you as if the first words and the last were at low volume and the conversation in the middle sound good, any had that problem?) So, internal calls sound good between 841's but sound volume is weird at the start and end of a sentence. Calling the outside lines via E1's, I can listen to people without problems but they heard me as choppy or cut off. Anybody had issues like this? Is it asterisk or the phones or what? Hope you can help Guys, Im really banging my head against the wall here. ___ --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 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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] Caller Name: Asterisk reading too fast
Have you tried placing a Wait(1) before Answer() in your dialplan? On Friday 16 September 2005 11:23, J Thomas wrote: I asked my telco to release caller name on the PRI. Earlier they were releasing only the phone number. I still did not see the name, but only the number in caller id. Actually I now see number twice. When I inquired with them this is the response I got: I ran a trace on your TG. I see that your switch is picking up the call so fast that it is not able to pick up the name. The name is being sent, but I suspect after it is too late. This is something that will need to be corrected in your switch. I have attached a sample call out of the trace I performed this morning. They have sent me the trace file. Is there a way as it is in Asterisk so that it reads the caller name properly? Thanks, -- jt ___ --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 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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] Newman Telecom files
This is happening because it is using the caller id struct members for CVS HEAD instead of the ones for v1-0. If you send the file to me I can fix it in a few minutes for you or you could try going to the lines where you get the error change each appearance of cid.xxx for its corresponding value in the ast_channel struct. For example: cid.cid_dnid -- dnid PS. Not all of the fields can be mapped so easily. On Fri September 9 2005 20:41, Carlos Alperin wrote: I tried to compile CVS 1.0.9 adding the Newman Telecom routines NVFaxDetect NVBackgroundDetect, but as I only get the .c files and not the dynamic libraries, all that I get is: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-v1-0-09/08/05-16:03:13\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -c -o app_nv_faxdetect.o app_nv_faxdetect.c app_nv_faxdetect.c: In function `nv_detectfax_exec': app_nv_faxdetect.c:218: error: structure has no member named `cid' app_nv_faxdetect.c:235: error: structure has no member named `cid' app_nv_faxdetect.c:273: error: structure has no member named `cid' make[1]: *** [app_nv_faxdetect.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 When I run make on /usr/src/asterisk. I tried to build the libraries but I got the same errors app_nv_faxdetect.c:218: error: structure has no member named `cid' app_nv_faxdetect.c:235: error: structure has no member named `cid' app_nv_faxdetect.c:273: error: structure has no member named `cid' on the compilation. Some has any clue about this? Of course, I sent e-mails to [EMAIL PROTECTED] but I never got any answer about the libraries, and I found only the .c files. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Schreiter Sent: Thursday, September 08, 2005 11:14 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Pass through of T.38 Hi, I found some contradicting infos about pass through of T.38 data. Are there any experiences of just passing T.38 via SIP from one T.38 application or gateway trough asterisk to another T.38 application or gateway? Would asterisk maybe even pass T.38 from chan_oh323 to chan_sip (without understanding the content)? Please tell me, if you have knowledges or experiences on this topic! Othervice, and if I won't find further reliable information saying it cannot work, I'll try it. And of course I will report the results later here. Roger. -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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] Low handset microphone volume with Sipura SPA-841
Just in case somebody else has this problem, it seems that there is a bug in the 3.1.3a version of firmware of the Sipura SPA-841. Updating to the 3.1.4a version of the firmware solved the problem. On Sun August 28 2005 01:55, Juan Jose Comellas wrote: I have just bought several Sipura SPA-841 SIP phones, and after some testing I have found out that the volume received by other parties when calling using the handset is very low. I've been able to reproduce this problem in the 3 phones I've tested so far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem? There are only two holes for the microphone in the handset and they are really small. I was thinking that myabe this is the cause. Any thoughts? -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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] Is LDAPget module stable enough for enterprise usage?
You may want to check another LDAP search module for Asterisk which should scale much better than LDAPGet. You can find it here: http://www.comellas.com.ar/app_ldap/app_ldap-0.1.0.tar.bz2 It was specifically designed for that amount of users. Please let me know what you think of it. On Sat August 27 2005 03:48, Liu, Wen wrote: Hi, all. I am building a SER+asterisk PBX airming at around 10k persons' usage. For authentication purpose I am in favor of ldap storage, while I am not sure the current ldap module for asterisk(0.9.9.2) is stable enough? sorry I do not master the proper testing mechanisms to find out myself. Thanks in advance. ___ --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 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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] Low handset microphone volume with Sipura SPA-841
The firmware on the phones is version 3.1.3(a). I will try today using the 3.1.4 firmware. The size of the display could be better, but the lack of a backlight is what really bothers me. On Sunday 28 August 2005 11:46, John Novack wrote: I have not experienced that problem, but earlier firmware resulted in an unusable speakerphone. Check if you have the latest firmware, then ask Sipura support for help. The one time I E-mailed them they were quite responsive. the 841 still has a worthless display though, doesn't it? Lack of backlightimg and too small isn't going to be fixed by a firmware change! John Novack Juan Jose Comellas wrote: I have just bought several Sipura SPA-841 SIP phones, and after some testing I have found out that the volume received by other parties when calling using the handset is very low. I've been able to reproduce this problem in the 3 phones I've tested so far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem? There are only two holes for the microphone in the handset and they are really small. I was thinking that myabe this is the cause. Any thoughts? ___ --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 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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] Low handset microphone volume with Sipura SPA-841
I tried changing the gain settings and also the volume settings in the User tab, Audio Volume section. I didn't notice any change in the microphone output volume. On Sunday 28 August 2005 18:20, Rob Lith wrote: In Admin/Advanced have you tried the Handset Input Gain: settings? Rob On 8/28/05, Juan Jose Comellas [EMAIL PROTECTED] wrote: I have just bought several Sipura SPA-841 SIP phones, and after some testing I have found out that the volume received by other parties when calling using the handset is very low. I've been able to reproduce this problem in the 3 phones I've tested so far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem? There are only two holes for the microphone in the handset and they are really small. I was thinking that myabe this is the cause. Any thoughts? -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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 ___ --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 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Low handset microphone volume with Sipura SPA-841
I have just bought several Sipura SPA-841 SIP phones, and after some testing I have found out that the volume received by other parties when calling using the handset is very low. I've been able to reproduce this problem in the 3 phones I've tested so far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem? There are only two holes for the microphone in the handset and they are really small. I was thinking that myabe this is the cause. Any thoughts? -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --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] Full T38 sip Faxing now Available
Please send this information to me also. On Thu July 28 2005 01:03, Michael D Schelin wrote: Hello everybody, for all of you that have searched for a real fax solution, look no further. We now have T38 faxing. Please contact me for more information. Thanks Michael D. Schelin ShellTel 626-814-2354 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel configuration for Argentina
I'm having some trouble dialing phone numbers in Argentina with Digium Zaptel cards. Does anyone have some sample configuration that works with Digium TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf and /etc/asterisk/zapata.conf. I have two Zaptel cards: the first one has 1 FXS port and 3 FXO ports; the second one has 4 FXO ports. My current configuration is the following: /etc/zaptel.conf loadzone=us defaultzone=us # The first Zaptel card has the FXS port. fxoks=1 fxsks=2-8 /etc/asterisk/zapata.conf [channels] rxgain=0.0 txgain=0.0 musiconhold=default busydetect=yes busycount=5 callprogress=yes echocancel=yes ; Internal FAX machine (FXS #1) signalling=fxo_ks language=en immediate=no callwaiting=yes context=pstn-outbound-fax channel = 1 ; Fax phone line (FXO #8) signalling=fxs_ks language=en group=2 callerid=asreceived context=pstn-inbound-fax channel = 8 ; Voice phone lines (FXO #2, #3, #4, #5, #6, #7) signalling=fxs_ks language=en callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes callgroup=1 pickupgroup=1 group=1 useincomingcalleridonzaptransfer=yes callerid=asreceived context=pstn-inbound-voice channel = 2-7 Thanks -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LDAP search application for Asterisk
I'm sending an Asterisk module I've written to see if anybody finds it useful or wants to provide some feedback. The name of the module is app_ldap and the application it provides is named LDAPSearch. LDAPSearch allows any kind of searches on an LDAP directory from the Asterisk dialplan. It returns its results using channel variables and it was specifically designed to be able to dial by name (users, contacts from an addressbook, etc.). One of its most interesting features is the extension it adds to the LDAP filter syntax (#= operator) to be able to match LDAP entries against a series of digits using the standard touch tone phone key mapping to make its comparisons. The syntax used by the application is the following: LDAPSearch(filter,[attr_1[:attr_2[:...]]][,max entries[,sort attr[,base DN[,scope) The documentation explaining its configuration and usage is included in doc/html/app_ldap.html. The only strange thing needed to build it is SCons (http://www.scons.org), which btw is an excellent build tool. You'll also need an LDAP client library and an LDAP server to test it. Everything has been tested using the OpenLDAP client library against both the OpenLDAP server and Microsoft Active Directory 2000. You can download it from: http://comellas.com.ar/app_ldap/app_ldap-0.1.0.tar.bz2 You'll need some knowledge of LDAP to be able to use this application successfully. Please report any problems you may have with it. -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP provider in Switzerland
This is a little bit off-topic, so forgive me. Does anybody know of any VoIP phone line provider in Switzerland that supports a VoIP protocol that can be connected with Asterisk? Thanks. -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Native MoH patch for 1.0.8?
I will be updating the native MOH patch to 1.0.8 during this week. I'll post it to the list once it's ready. On Monday 27 June 2005 08:46, Patrick wrote: Hi all, I was reading http://bugs.digium.com/view.php?id=2639 and it seems that anthm's great native MoH patch only works on HEAD. Does anyone have a version of the native MoH patch that works on 1.0.8? If so please point me to its location or email it off-list. Thanks and regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp in 64 bit Linux on AMD64
Is there any stable version of spandsp that works on a 64 bit Linux on an AMD64 machine. When compiling version 0.0.1k I get the following error: gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -c testcpuid.c -MT testcpuid.lo -MD -MP -MF .deps/testcpuid.TPlo -fPIC -DPIC -o .libs/testcpuid.lo /tmp/ccXxGHg6.s: Assembler messages: /tmp/ccXxGHg6.s:8: Error: suffix or operands invalid for `pushf' /tmp/ccXxGHg6.s:9: Error: suffix or operands invalid for `pushf' /tmp/ccXxGHg6.s:10: Error: suffix or operands invalid for `pop' /tmp/ccXxGHg6.s:13: Error: suffix or operands invalid for `push' /tmp/ccXxGHg6.s:14: Error: suffix or operands invalid for `popf' /tmp/ccXxGHg6.s:15: Error: suffix or operands invalid for `pushf' /tmp/ccXxGHg6.s:16: Error: suffix or operands invalid for `pop' /tmp/ccXxGHg6.s:17: Error: suffix or operands invalid for `popf' make[2]: *** [testcpuid.lo] Error 1 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conferencing without Meetme
I'm currently writing some code to support conferencing in Asterisk without using the Meetme application. The conference runs in its own thread and every new inbound or outbound channel that is created is passed to it. This thread runs the conference loop reading and writing frames to each channel. I'm writing this as if it were a bridge with more than two channels, and I'm not using the native bridging capabilities of the channels because apparently they only allow two channels. Is there any special precaution that I have to be aware of when doing this? Do I have to masquerade the channels that are inserted into the conference? The channels will mainly use SIP (maybe IAX2 too occasionally). Thanks for your help. -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice + DTMF
Asterisk can do this easily, but I don't know if there's any application that will allow you to do this (for your specific needs) in the dialplan. If you write your own C module for this purpose you'll have no problem implementing this functionality. On Friday 19 November 2004 15:04, Geraldo Fco. do Espírito Santo Jr. wrote: Hi everyone, I am working in a project, I would like some advices. We will have a Call Centers Agent that will call a mobile phone connect to a computer in a car, the driver can talk (live voice) and the computer can receive/send DTMF digits (codes) while a conversation. A)Is it possible to * recognize this DTMF digits during the conversation? B) Can a softphone recognize this DTMF digits during the conversation? C) How difficult it is? Thanks, Gerald -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ 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] Dynamic dialplan
We intend to use Asterisk with a very large dialplan (with a lot of functionality for 3000+ users). Each user will be able to change several of his parameters in the dialplan, so we will be forced to reload the diaplan constantly. Has anybody else any previous experience with a similar installation? There are some things that we'd like to know, if anybody can help us. These are: - Is this something that can be done safely with Asterisk? - Can we have a diaplan configuration update every 5 or 10 minutes without service interruption? - What happens to new calls while the dialplan configuration is being reloaded? - What happens to active calls after the dialplan configuration is updated? - Can we do partial updates of the dialplan (e.g. update a specific context instead of the whole dialplan configuration)? - Can Asterisk have its dialplan in a database instead of having it always in memory? Thanks. -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ 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
