Re: [asterisk-users] PRI: sometimes Asterisk drop calls
On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: I'm interested to understand why I many messages like: WARNING[21314] chan_zap.c: Ring requested on channel 0/1 already in use on span 1. Hanging up owner How can a channel be already in use??? That means the channel is busy...if it is so then it is all right...but maybe that shouldn't be a warning but a notice or something else...should it? What this means is that a call has arrived from your Telco on a Zap channel. Asterisk THINKS that this channel is already in use - In my case, asterisk is wrong. Sadly, asterisk hangs up the new call (so no-one can call you anymore!) and does not have an option to clean up the call data to show the channel as free. Outbound calls are fine, but will never be placed on this hung channel. IMHO, 99% of the time, if the telco says a channel is free, they know best!!! As per my other email, I see this problem on one site. Sadly I have not been able to identify where it comes from (although I suspect broken SIP clients upsetting Asterisk) What versions of zaptel/asterisk do you use? What hardware, and what phone devices? Perhaps people with this problem have something in common. To start the ball rolling: Software: Zaptel 1.0.9, Asterisk 1.0.9, BRIstuff-0.2.0, wanpipe-2.3.2 PRI interface: Sangoma A101U (UK E1) Phones on sites with NO problems: snom, elmeg, Aastra, Linksys/Sipura Phones on problem site: Hitachi WIP3000, Zyxel F1000 (?) Regards, Steve ___ --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] PRI: sometimes Asterisk drop calls
Hi Steve, I agree with you..telco knows better! If telco sends a ring on channel X and asterisk has already used it, couldn't asterisk shift that call on another channel Y or it is obliged to answer on channel X? In other words, if asterisk get a ring on channel 3 and channel 3 is in use, Asterisk should use channel 6 to answer the call and so connect channel 3 to 6. Isn't it? Or I have not correctly understood how telco and Asterisk speak between them? My PBX configuration is (may change due to upgrades): Debian Sarge 3.1 r0a asterisk-install: v6 Asterisk: 1.2.9.1 Zaptel: 1.2.6 Libpri: 1.2.3 BRI: install-misdn-mqueue 0.3.1-rc23 Giorgio Incantalupo Steve Davies wrote: On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: I'm interested to understand why I many messages like: WARNING[21314] chan_zap.c: Ring requested on channel 0/1 already in use on span 1. Hanging up owner How can a channel be already in use??? That means the channel is busy...if it is so then it is all right...but maybe that shouldn't be a warning but a notice or something else...should it? What this means is that a call has arrived from your Telco on a Zap channel. Asterisk THINKS that this channel is already in use - In my case, asterisk is wrong. Sadly, asterisk hangs up the new call (so no-one can call you anymore!) and does not have an option to clean up the call data to show the channel as free. Outbound calls are fine, but will never be placed on this hung channel. IMHO, 99% of the time, if the telco says a channel is free, they know best!!! As per my other email, I see this problem on one site. Sadly I have not been able to identify where it comes from (although I suspect broken SIP clients upsetting Asterisk) What versions of zaptel/asterisk do you use? What hardware, and what phone devices? Perhaps people with this problem have something in common. To start the ball rolling: Software: Zaptel 1.0.9, Asterisk 1.0.9, BRIstuff-0.2.0, wanpipe-2.3.2 PRI interface: Sangoma A101U (UK E1) Phones on sites with NO problems: snom, elmeg, Aastra, Linksys/Sipura Phones on problem site: Hitachi WIP3000, Zyxel F1000 (?) Regards, Steve ___ --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] PRI: sometimes Asterisk drop calls
On 9/13/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi Steve, I agree with you..telco knows better! If telco sends a ring on channel X and asterisk has already used it, couldn't asterisk shift that call on another channel Y or it is obliged to answer on channel X? The telco is in charge and is not asking, it is telling asterisk that there is a call on channel X. If asterisk thinks this channel is busy, then it is wrong because it must agree that a channel is busy (SETUP/SETUP_ACK) with the telco first. Probably, a now finished call was made on that channel, and some old data in asterisk has not been cleaned-up when the call went away. In other words, if asterisk get a ring on channel 3 and channel 3 is in use, Asterisk should use channel 6 to answer the call and so connect channel 3 to 6. Isn't it? Or I have not correctly understood how telco and Asterisk speak between them? BRI/PRI channels are like point to point data pipes, they cannot be cross connected in that way. If the telco knows that channel 3 is busy and channel 6 is available, it will just use channel 6 in the first place. (I stand ready to be told I am wrong here) My PBX configuration is (may change due to upgrades): Debian Sarge 3.1 r0a asterisk-install: v6 Asterisk: 1.2.9.1 Zaptel: 1.2.6 Libpri: 1.2.3 BRI: install-misdn-mqueue 0.3.1-rc23 No obvious common factor then... Steve ___ --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] PRI: sometimes Asterisk drop calls
Problema solved! Just put resetinterval=never inside zapata.conf Giorgio Incantalupo Giorgio Incantalupo wrote: Hi, I installed an Asterisk box with a sangoma A102 PRI card. Sometimes Asterisk drops calls...there is nothing inside logs but these warnings: Sep 11 15:00:18 WARNING[3503] chan_zap.c: Ring requested on channel 0/6 already in use on span 1. Hanging up owner. Sep 11 15:00:22 WARNING[3503] chan_zap.c: Got restart ack on channel 0/6 span 1 with owner Sep 11 15:00:30 WARNING[3503] chan_zap.c: Ring requested on channel 0/3 already in use on span 1. Hanging up owner. Sep 11 15:29:38 WARNING[3497] chan_sip.c: Maximum retries exceeded on transmission [EMAIL PROTECTED] .168.3.175 for seqno 2 (Critical Response) Sep 11 15:30:04 WARNING[3497] chan_sip.c: Maximum retries exceeded on transmission [EMAIL PROTECTED] 2.168.3.175 for seqno 2 (Critical Response) Sep 11 15:30:24 WARNING[3497] chan_sip.c: Maximum retries exceeded on transmission [EMAIL PROTECTED] 2.168.3.175 for seqno 2 (Critical Response) Sep 11 15:31:34 WARNING[3503] chan_zap.c: Ring requested on channel 0/6 already in use on span 1. Hanging up owner. Sep 11 16:00:08 WARNING[3503] chan_zap.c: Ring requested on channel 0/6 already in use on span 1. Hanging up owner. Sep 11 16:00:26 WARNING[3503] chan_zap.c: Got restart ack on channel 0/6 span 1 with owner Sep 11 16:03:13 WARNING[3503] chan_zap.c: Ring requested on channel 0/6 already in use on span 1. Hanging up owner. Sep 11 16:03:15 WARNING[15530] app_dial.c: Unable to forward voice Sep 11 16:07:09 WARNING[3503] chan_zap.c: Ring requested on channel 0/6 already in use on span 1. Hanging up owner. Sep 11 16:13:22 WARNING[15925] app_dial.c: Unable to forward voice Sep 11 16:15:37 WARNING[3503] chan_zap.c: Ring requested on channel 0/6 already in use on span 1. Hanging up owner. Sep 11 16:21:26 WARNING[3504] chan_zap.c: Call specified, but not found? Sep 11 16:23:21 WARNING[16267] app_dial.c: Unable to forward Anyone ever got these messages? What do they mean? How can I fix them? TIA Giorgio Incantalupo ___ --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] PRI: sometimes Asterisk drop calls
For the curious, can anyone tell me how this flag fixes the issue? - I have seen the error before, but always assumed it was related to hung channels. Thanks, Steve On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Problema solved! Just put resetinterval=never inside zapata.conf Giorgio Incantalupo Giorgio Incantalupo wrote: Hi, I installed an Asterisk box with a sangoma A102 PRI card. Sometimes Asterisk drops calls...there is nothing inside logs but these warnings: Sep 11 15:00:18 WARNING[3503] chan_zap.c: Ring requested on channel 0/6 already in use on span 1. Hanging up owner. Sep 11 15:00:22 WARNING[3503] chan_zap.c: Got restart ack on channel 0/6 span 1 with owner Sep 11 15:00:30 WARNING[3503] chan_zap.c: Ring requested on channel 0/3 already in use on span 1. Hanging up owner. Sep 11 15:29:38 WARNING[3497] chan_sip.c: Maximum retries exceeded on transmission [EMAIL PROTECTED] .168.3.175 for seqno 2 (Critical Response) Sep 11 15:30:04 WARNING[3497] chan_sip.c: Maximum retries exceeded on transmission [EMAIL PROTECTED] 2.168.3.175 for seqno 2 (Critical Response) Sep 11 15:30:24 WARNING[3497] chan_sip.c: Maximum retries exceeded on transmission [EMAIL PROTECTED] 2.168.3.175 for seqno 2 (Critical Response) Sep 11 15:31:34 WARNING[3503] chan_zap.c: Ring requested on channel 0/6 already in use on span 1. Hanging up owner. Sep 11 16:00:08 WARNING[3503] chan_zap.c: Ring requested on channel 0/6 already in use on span 1. Hanging up owner. Sep 11 16:00:26 WARNING[3503] chan_zap.c: Got restart ack on channel 0/6 span 1 with owner Sep 11 16:03:13 WARNING[3503] chan_zap.c: Ring requested on channel 0/6 already in use on span 1. Hanging up owner. Sep 11 16:03:15 WARNING[15530] app_dial.c: Unable to forward voice Sep 11 16:07:09 WARNING[3503] chan_zap.c: Ring requested on channel 0/6 already in use on span 1. Hanging up owner. Sep 11 16:13:22 WARNING[15925] app_dial.c: Unable to forward voice Sep 11 16:15:37 WARNING[3503] chan_zap.c: Ring requested on channel 0/6 already in use on span 1. Hanging up owner. Sep 11 16:21:26 WARNING[3504] chan_zap.c: Call specified, but not found? Sep 11 16:23:21 WARNING[16267] app_dial.c: Unable to forward Anyone ever got these messages? What do they mean? How can I fix them? ___ --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] PRI: sometimes Asterisk drop calls
Steve Davies wrote: For the curious, can anyone tell me how this flag fixes the issue? - I have seen the error before, but always assumed it was related to hung channels. Thanks, Steve On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Problema solved! Just put resetinterval=never inside zapata.conf Giorgio Incantalupo If memory serves correctly, I believe the parameter was added a couple of years ago as a means / workaround for hung channels. At the time, there was not any overwhelming evidence as why a channel would occasionally hang. Some of the possibilities included unusual interaction from the opposite end of the T1/E1, anomalies in the dialplan, etc. Now that a substantial amount of work / changes have been made relative to PRI's and other internal asterisk code, there appears to be less of a need to reset. A reasonable approach might be to apply the parameter and pay close attention to channels that might be in some strange state. If none are observed, then leave it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI: sometimes Asterisk drop calls
On 9/12/06, Rich Adamson [EMAIL PROTECTED] wrote: Steve Davies wrote: For the curious, can anyone tell me how this flag fixes the issue? - I have seen the error before, but always assumed it was related to hung channels. Thanks, Steve On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Problema solved! Just put resetinterval=never inside zapata.conf Giorgio Incantalupo If memory serves correctly, I believe the parameter was added a couple of years ago as a means / workaround for hung channels. At the time, there was not any overwhelming evidence as why a channel would occasionally hang. Some of the possibilities included unusual interaction from the opposite end of the T1/E1, anomalies in the dialplan, etc. Now that a substantial amount of work / changes have been made relative to PRI's and other internal asterisk code, there appears to be less of a need to reset. A reasonable approach might be to apply the parameter and pay close attention to channels that might be in some strange state. If none are observed, then leave it. Thanks for that. I have a customer who is using Asterisk 1.0.x, and I am tempted to backport this fix from the 1.2.x code where it was introduced. Cheers, Steve ___ --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] PRI: sometimes Asterisk drop calls
Steve Davies wrote: On 9/12/06, Rich Adamson [EMAIL PROTECTED] wrote: Steve Davies wrote: For the curious, can anyone tell me how this flag fixes the issue? - I have seen the error before, but always assumed it was related to hung channels. Thanks, Steve On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Problema solved! Just put resetinterval=never inside zapata.conf Giorgio Incantalupo If memory serves correctly, I believe the parameter was added a couple of years ago as a means / workaround for hung channels. At the time, there was not any overwhelming evidence as why a channel would occasionally hang. Some of the possibilities included unusual interaction from the opposite end of the T1/E1, anomalies in the dialplan, etc. Now that a substantial amount of work / changes have been made relative to PRI's and other internal asterisk code, there appears to be less of a need to reset. A reasonable approach might be to apply the parameter and pay close attention to channels that might be in some strange state. If none are observed, then leave it. Thanks for that. I have a customer who is using Asterisk 1.0.x, and I am tempted to backport this fix from the 1.2.x code where it was introduced. From a personal perspective, I think I'd hold off on the back port and devote that time towards testing the soon to be released version (now in Trunk). If you've watched the number and type of changes that have gone into SVN Trunk in the last couple of months, it appears as though a significant number of possible memory leaks, sip code, infrastructure code, PRI code changes, etc, have been applied that would be beneficial for all production systems. There also appears to be a fair amount of work that will be needed to upgrade dialplan syntax (etc) for the new release. Best guess is that once the Trunk code gets past the beta testing phase, it will likely be the asterisk code of choice for most/all production systems. Consider the above is only my $0.02 worth. ;) ___ --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] PRI: sometimes Asterisk drop calls
Hi, thanks to all I solved the calls dropped problem, it was resetinterval parameter in zapata.now asterisk does not drop calls anymore. I do not get the message: WARNING[3503] chan_zap.c: Got restart ack on channel 0/6 span 1 with owner anymore...but I get all the others. I'm interested to understand why I many messages like: WARNING[21314] chan_zap.c: Ring requested on channel 0/1 already in use on span 1. Hanging up owner How can a channel be already in use??? That means the channel is busy...if it is so then it is all right...but maybe that shouldn't be a warning but a notice or something else...should it? TIA Giorgio Incantalupo Rich Adamson wrote: Steve Davies wrote: On 9/12/06, Rich Adamson [EMAIL PROTECTED] wrote: Steve Davies wrote: For the curious, can anyone tell me how this flag fixes the issue? - I have seen the error before, but always assumed it was related to hung channels. Thanks, Steve On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Problema solved! Just put resetinterval=never inside zapata.conf Giorgio Incantalupo If memory serves correctly, I believe the parameter was added a couple of years ago as a means / workaround for hung channels. At the time, there was not any overwhelming evidence as why a channel would occasionally hang. Some of the possibilities included unusual interaction from the opposite end of the T1/E1, anomalies in the dialplan, etc. Now that a substantial amount of work / changes have been made relative to PRI's and other internal asterisk code, there appears to be less of a need to reset. A reasonable approach might be to apply the parameter and pay close attention to channels that might be in some strange state. If none are observed, then leave it. Thanks for that. I have a customer who is using Asterisk 1.0.x, and I am tempted to backport this fix from the 1.2.x code where it was introduced. From a personal perspective, I think I'd hold off on the back port and devote that time towards testing the soon to be released version (now in Trunk). If you've watched the number and type of changes that have gone into SVN Trunk in the last couple of months, it appears as though a significant number of possible memory leaks, sip code, infrastructure code, PRI code changes, etc, have been applied that would be beneficial for all production systems. There also appears to be a fair amount of work that will be needed to upgrade dialplan syntax (etc) for the new release. Best guess is that once the Trunk code gets past the beta testing phase, it will likely be the asterisk code of choice for most/all production systems. Consider the above is only my $0.02 worth. ;) ___ --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
