Re: [asterisk-users] libpri : Q.931 Called Party Number interpreted as empty
2011/11/2 giovanni.v i...@keybits.org On 02/11/2011 18.45, Olivier wroteo: 1. As the above line comes from libpri, how can one be certain the telco side didn't send the weird NUL byte ? Assuming libpri debug doesn't mess nothing is quite sure the NUL come in from the telco. So, would you still rate this as a libpri bug or a Telco issue ? I would say if this is confirmed to come from the Telco network, then this is a Telco issue. 2. Anyway, as Q.931 allows any IA5 (ISO 646) character, libpri should notify sysadmin for every non-IA5 character received But NUL is an IA5 character, so I think must be handled like any other one. - http://skew.org/iso-ir-001/ Then, how should such a valid but unuseful character, if I may rate it as such, be handed, then ? To me: A. we should not expect any called number to include any character but those in the 0 to 9 range. B. we should notify sysadmin anytime an unuseful character is received (to let him, for instance, ask telco do whatever is needed for this to stop). -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] State of Asterisk+Virtualization+Timing
On Tue, 2011-11-01 at 12:08 -0500, Tim Nelson wrote: Greetings- I'm about to dive into the process of virtualizing some of my Asterisk (primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, the primary issue preventing me from moving was the lack of proper timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either OpenVZ or KVM, but each seem to have independent issues that need to be addressed: OpenVZ - Better resource usage, lower overhead. Primary issue is how to grant access to host node timing source (physical device, or dahdi_dummy in /dev/dahdi/) to the containerized Asterisk process. KVM - Higher overhead, easier installation, 'true virtualization'. Primary issue is not timing per se, but KVM scheduling. Timing source, while present from dahdi_dummy natively may still not get proper scheduling by KVM process. This could also affect general call quality (even non IAX2 trunked voice), DTMF, etc. I have to believe there are others running virtualized Asterisk installations with some degree of success on OpenVZ or KVM. Care to share your thoughts? You mist out one more mature virtualization technique: XEN Virtual machines can use both hardware- or paravirtualization. I have used both asterisk (1.4, 1.6.x and now 1.8) to separate machines where people should do their sip-registration (internet / intranet / pstn-gateway) and the actual dial-beast. Main advantage for virtualization is (besides easy scaling) that you can perform an upgrade in no-time (one VM-machine down, other up) Don't like it: back in seconds! Migration with an asterisk on real hardware takes much more resources. Both in iron and in time. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] libpri : Q.931 Called Party Number interpreted as empty
On 03/11/2011 9.08, Olivier wrote: Then, how should such a valid but unuseful character, if I may rate it as such, be handed, then ? To me: A. we should not expect any called number to include any character but those in the 0 to 9 range. B. we should notify sysadmin anytime an unuseful character is received (to let him, for instance, ask telco do whatever is needed for this to stop). Oliver, I partially agree with you but especially when standards are involved need to play more like devil's advocate. About A: 1 - Q.931 is not only isdn (gsm, ss7...), usually a specification are built to plug-in and play together with other specifications. If a standard says to accept IA-5 that must be done, else we may break something else. 2 - Maybe the telco switch violate some other specification about the called number format but: 2a) because of 1) discarding the whole data isn't the right solution. 2b) I don't know if a country specific rule allow for that format. So totally agree in B from you and the solution suggested by Mudgett: discard chars that we can't handle log the event so the admin will be able to do further investigations. p.s.: sorry for my /Spaghetti-English-language/. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] duration limits in Dial() not being enforced at correct time
Hi, We're trying to time-limit some calls by specifying L(x:y:z) as an option to the Dial command. If we set the limit to a fairly short duration (eg 120 seconds) then Asterisk seems to issue the hangup at about the right time. However, for longish calls we're seeing quite a bit of overspill. For example we tried to limit one to 1338 seconds but Asterisk didn't hang up until 1384 seconds after the call was answered. Also, the error is not always consistent - a second test call also limited to 1338 seconds was hung up by Asterisk after 1347 seconds. We saw this problem with Asterisk 1.6 but we've now tried on Asterisk 1.8.6.0 and are having the same problem. Here's a log from the Asterisk 1.8.6.0 box for the test call that should have been limited to 1338 seconds but was actually ended after 1384 seconds. The server wasn't carrying any other calls at the time or doing anything else so the load would have been very low. [Nov 2 16:47:37] VERBOSE[2029] pbx.c: -- Executing [01476292501@service_nts_v2:57] Dial(DAHDI/i2/7622323283-4, DAHDI/g1/08451238347,,L(1338000:3:5000)M(service-nts-v2-register-answer)) in new stack [Nov 2 16:47:37] VERBOSE[2029] features.c: Limit Data for this call: [Nov 2 16:47:37] VERBOSE[2029] features.c: timelimit = 1338000 ms (1338.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: play_warning = 3 ms (30.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: play_to_caller = yes [Nov 2 16:47:37] VERBOSE[2029] features.c: play_to_callee = no [Nov 2 16:47:37] VERBOSE[2029] features.c: warning_freq = 5000 ms (5.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: start_sound= [Nov 2 16:47:37] VERBOSE[2029] features.c: warning_sound = /var/lib/asterisk/sounds/bespoke/beep_200ms [Nov 2 16:47:37] VERBOSE[2029] features.c: end_sound = [Nov 2 16:47:37] VERBOSE[2029] sig_pri.c: -- Requested transfer capability: 0x00 - SPEECH [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- Called DAHDI/g1/08451238347 [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 is proceeding passing it to DAHDI/i2/7622323283-4 [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 is ringing [Nov 2 16:47:38] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 answered DAHDI/i2/7622323283-4 [Nov 2 16:47:38] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:1] NoOp(DAHDI/i1/08451238347-3, ANSWER MACRO) in new stack [Nov 2 16:47:38] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:2] AGI(DAHDI/i1/08451238347-3, agi://127.0.0.1:4573/ServiceNTSV2,mode=answered,uniqueID=1320252457.17_1,uniqueIDB=1320252457.18,ddi=08451238347,Goto=agiOK1) in new stack [Nov 2 16:47:39] VERBOSE[2029] res_agi.c: -- AGI Script Executing Application: (Goto) Options: (agiOK1) [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Goto (macro-service-nts-v2-register-answer,s,7) [Nov 2 16:47:39] VERBOSE[2029] res_agi.c: -- DAHDI/i1/08451238347-3AGI Script agi://127.0.0.1:4573/ServiceNTSV2 completed, returning 0 [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:7] GotoIf(DAHDI/i1/08451238347-3, 1?agiOK2) in new stack [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Goto (macro-service-nts-v2-register-answer,s,13) [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:13] NoOp(DAHDI/i1/08451238347-3, register-answer macro finished) in new stack [Nov 2 16:47:39] VERBOSE[2029] chan_dahdi.c: -- Native bridging DAHDI/i2/7622323283-4 and DAHDI/i1/08451238347-3 [Nov 2 17:10:42] VERBOSE[2029] pbx.c: -- Executing [h@service_nts_v2:1] NoOp(DAHDI/i2/7622323283-4, number HANGING UP ... CHANNEL=DAHDI/i2/7622323283-4, channel1=1320252457.17_1, channel2=, HANGUPCAUSE=16, UNIQUEID=1320252457.17) in new stack Is this a known problem and are there any workarounds? -- Cheers, Kingsley. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] libpri : Q.931 Called Party Number interpreted as empty
2011/11/3 giovanni.v i...@keybits.org p.s.: sorry for my /Spaghetti-English-language/. Don't worry about your english : I'm not pround of mine ;-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration limits in Dial() not being enforced at correct time
Please elaborate on your flavor of DAHDI and LIBPRI and what type of DAHDI service you are using (PSTN, T1, etc). Speaking from a POTS line point of view, there can easily be a 7-10 second delay in the processing of DAHDI information (which would make your 1347 second call within tolerance). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart Sent: Thursday, November 03, 2011 5:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] duration limits in Dial() not being enforced at correct time Hi, We're trying to time-limit some calls by specifying L(x:y:z) as an option to the Dial command. If we set the limit to a fairly short duration (eg 120 seconds) then Asterisk seems to issue the hangup at about the right time. However, for longish calls we're seeing quite a bit of overspill. For example we tried to limit one to 1338 seconds but Asterisk didn't hang up until 1384 seconds after the call was answered. Also, the error is not always consistent - a second test call also limited to 1338 seconds was hung up by Asterisk after 1347 seconds. We saw this problem with Asterisk 1.6 but we've now tried on Asterisk 1.8.6.0 and are having the same problem. Here's a log from the Asterisk 1.8.6.0 box for the test call that should have been limited to 1338 seconds but was actually ended after 1384 seconds. The server wasn't carrying any other calls at the time or doing anything else so the load would have been very low. [Nov 2 16:47:37] VERBOSE[2029] pbx.c: -- Executing [01476292501@service_nts_v2:57] Dial(DAHDI/i2/7622323283-4, DAHDI/g1/08451238347,,L(1338000:3:5000)M(service-nts-v2-register-answer )) in new stack [Nov 2 16:47:37] VERBOSE[2029] features.c: Limit Data for this call: [Nov 2 16:47:37] VERBOSE[2029] features.c: timelimit = 1338000 ms (1338.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: play_warning = 3 ms (30.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: play_to_caller = yes [Nov 2 16:47:37] VERBOSE[2029] features.c: play_to_callee = no [Nov 2 16:47:37] VERBOSE[2029] features.c: warning_freq = 5000 ms (5.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: start_sound= [Nov 2 16:47:37] VERBOSE[2029] features.c: warning_sound = /var/lib/asterisk/sounds/bespoke/beep_200ms [Nov 2 16:47:37] VERBOSE[2029] features.c: end_sound = [Nov 2 16:47:37] VERBOSE[2029] sig_pri.c: -- Requested transfer capability: 0x00 - SPEECH [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- Called DAHDI/g1/08451238347 [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 is proceeding passing it to DAHDI/i2/7622323283-4 [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 is ringing [Nov 2 16:47:38] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 answered DAHDI/i2/7622323283-4 [Nov 2 16:47:38] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:1] NoOp(DAHDI/i1/08451238347-3, ANSWER MACRO) in new stack [Nov 2 16:47:38] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:2] AGI(DAHDI/i1/08451238347-3, agi://127.0.0.1:4573/ServiceNTSV2,mode=answered,uniqueID=1320252457.17_1,un iqueIDB=1320252457.18,ddi=08451238347,Goto=agiOK1) in new stack [Nov 2 16:47:39] VERBOSE[2029] res_agi.c: -- AGI Script Executing Application: (Goto) Options: (agiOK1) [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Goto (macro-service-nts-v2-register-answer,s,7) [Nov 2 16:47:39] VERBOSE[2029] res_agi.c: -- DAHDI/i1/08451238347-3AGI Script agi://127.0.0.1:4573/ServiceNTSV2 completed, returning 0 [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:7] GotoIf(DAHDI/i1/08451238347-3, 1?agiOK2) in new stack [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Goto (macro-service-nts-v2-register-answer,s,13) [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:13] NoOp(DAHDI/i1/08451238347-3, register-answer macro finished) in new stack [Nov 2 16:47:39] VERBOSE[2029] chan_dahdi.c: -- Native bridging DAHDI/i2/7622323283-4 and DAHDI/i1/08451238347-3 [Nov 2 17:10:42] VERBOSE[2029] pbx.c: -- Executing [h@service_nts_v2:1] NoOp(DAHDI/i2/7622323283-4, number HANGING UP ... CHANNEL=DAHDI/i2/7622323283-4, channel1=1320252457.17_1, channel2=, HANGUPCAUSE=16, UNIQUEID=1320252457.17) in new stack Is this a known problem and are there any workarounds? -- Cheers, Kingsley. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks
Hi, I'm still strugling with my CallerID presentation problem. Let me remind it : My setup is: Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob cellphone Ive configured Asterisk so that whenever Bob forwards its incoming call to its cellphone, the later phone should present Alice's number. I was originally told that sometimes Bob would be presented Alice's number, sometimes the dialed number (which in this case, also match the ANI or the receptionnist ID). Now, I can't certify this ever happened : maybe, it did happen, maybe not. What I can certify is this: 1. out of 32 different callers, 20 callers are presented with the correct number and 12 with the dialed number, 2. all those 32 callers are cellphones operated by the same (incumbent) telco which also operates ISDN, 3. Bob's cellphone is also operated by the same (incumbent) telco, 4. all this tries were done the same day, one after the other. The best explanation I can think of is this: Depending on the route used, the ANI is used instead of the presented caller ID. To prove that, I'll try to record 2 calls for the same caller and toward the same destination: one with the awaited presentation, one with a wrong one. (Sending this to the telco and have them change anything is an other story). Comments and suggestions are welcome. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks
What version of Asterisk? Is the forwarding done using Followme, attended transfer or blind transfer? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, November 03, 2011 8:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks Hi, I'm still strugling with my CallerID presentation problem. Let me remind it : My setup is: Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob cellphone Ive configured Asterisk so that whenever Bob forwards its incoming call to its cellphone, the later phone should present Alice's number. I was originally told that sometimes Bob would be presented Alice's number, sometimes the dialed number (which in this case, also match the ANI or the receptionnist ID). Now, I can't certify this ever happened : maybe, it did happen, maybe not. What I can certify is this: 1. out of 32 different callers, 20 callers are presented with the correct number and 12 with the dialed number, 2. all those 32 callers are cellphones operated by the same (incumbent) telco which also operates ISDN, 3. Bob's cellphone is also operated by the same (incumbent) telco, 4. all this tries were done the same day, one after the other. The best explanation I can think of is this: Depending on the route used, the ANI is used instead of the presented caller ID. To prove that, I'll try to record 2 calls for the same caller and toward the same destination: one with the awaited presentation, one with a wrong one. (Sending this to the telco and have them change anything is an other story). Comments and suggestions are welcome. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration limits in Dial() not being enforced at correct time
On Thu, Nov 3, 2011 at 18:44, Danny Nicholas da...@debsinc.com wrote: Please elaborate on your flavor of DAHDI and LIBPRI and what type of DAHDI service you are using (PSTN, T1, etc). Speaking from a POTS line point of view, there can easily be a 7-10 second delay in the processing of DAHDI information (which would make your 1347 second call within tolerance). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart Sent: Thursday, November 03, 2011 5:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] duration limits in Dial() not being enforced at correct time Hi, We're trying to time-limit some calls by specifying L(x:y:z) as an option to the Dial command. If we set the limit to a fairly short duration (eg 120 seconds) then Asterisk seems to issue the hangup at about the right time. However, for longish calls we're seeing quite a bit of overspill. For example we tried to limit one to 1338 seconds but Asterisk didn't hang up until 1384 seconds after the call was answered. Also, the error is not always consistent - a second test call also limited to 1338 seconds was hung up by Asterisk after 1347 seconds. We saw this problem with Asterisk 1.6 but we've now tried on Asterisk 1.8.6.0 and are having the same problem. Here's a log from the Asterisk 1.8.6.0 box for the test call that should have been limited to 1338 seconds but was actually ended after 1384 seconds. The server wasn't carrying any other calls at the time or doing anything else so the load would have been very low. [Nov 2 16:47:37] VERBOSE[2029] pbx.c: -- Executing [01476292501@service_nts_v2:57] Dial(DAHDI/i2/7622323283-4, DAHDI/g1/08451238347,,L(1338000:3:5000)M(service-nts-v2-register-answer )) in new stack [Nov 2 16:47:37] VERBOSE[2029] features.c: Limit Data for this call: [Nov 2 16:47:37] VERBOSE[2029] features.c: timelimit = 1338000 ms (1338.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: play_warning = 3 ms (30.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: play_to_caller = yes [Nov 2 16:47:37] VERBOSE[2029] features.c: play_to_callee = no [Nov 2 16:47:37] VERBOSE[2029] features.c: warning_freq = 5000 ms (5.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: start_sound= [Nov 2 16:47:37] VERBOSE[2029] features.c: warning_sound = /var/lib/asterisk/sounds/bespoke/beep_200ms [Nov 2 16:47:37] VERBOSE[2029] features.c: end_sound = [Nov 2 16:47:37] VERBOSE[2029] sig_pri.c: -- Requested transfer capability: 0x00 - SPEECH [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- Called DAHDI/g1/08451238347 [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 is proceeding passing it to DAHDI/i2/7622323283-4 [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 is ringing [Nov 2 16:47:38] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 answered DAHDI/i2/7622323283-4 [Nov 2 16:47:38] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:1] NoOp(DAHDI/i1/08451238347-3, ANSWER MACRO) in new stack [Nov 2 16:47:38] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:2] AGI(DAHDI/i1/08451238347-3, agi:// 127.0.0.1:4573/ServiceNTSV2,mode=answered,uniqueID=1320252457.17_1,un iqueIDB=1320252457.18,ddi=08451238347,Goto=agiOK1) in new stack [Nov 2 16:47:39] VERBOSE[2029] res_agi.c: -- AGI Script Executing Application: (Goto) Options: (agiOK1) [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Goto (macro-service-nts-v2-register-answer,s,7) [Nov 2 16:47:39] VERBOSE[2029] res_agi.c: -- DAHDI/i1/08451238347-3AGI Script agi://127.0.0.1:4573/ServiceNTSV2 completed, returning 0 [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:7] GotoIf(DAHDI/i1/08451238347-3, 1?agiOK2) in new stack [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Goto (macro-service-nts-v2-register-answer,s,13) [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:13] NoOp(DAHDI/i1/08451238347-3, register-answer macro finished) in new stack [Nov 2 16:47:39] VERBOSE[2029] chan_dahdi.c: -- Native bridging DAHDI/i2/7622323283-4 and DAHDI/i1/08451238347-3 [Nov 2 17:10:42] VERBOSE[2029] pbx.c: -- Executing [h@service_nts_v2 :1] NoOp(DAHDI/i2/7622323283-4, number HANGING UP ... CHANNEL=DAHDI/i2/7622323283-4, channel1=1320252457.17_1, channel2=, HANGUPCAUSE=16, UNIQUEID=1320252457.17) in new stack Is this a known problem and are there any workarounds? -- Cheers, Kingsley. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live
Re: [asterisk-users] duration limits in Dial() not being enforced at correct time
If you dial to a Local/Context and use your time limits on that and then do your dial to your DAHDI device inside that context does that have any effect on the time limits working. We have used time limits with Local/Context dials and had them work with out any known issues. Thanks Bryant Zimmerman From: amit anand onewaytoconn...@gmail.com Sent: Thursday, November 03, 2011 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] duration limits in Dial() not being enforced at correct time On Thu, Nov 3, 2011 at 18:44, Danny Nicholas da...@debsinc.com wrote: Please elaborate on your flavor of DAHDI and LIBPRI and what type of DAHDI service you are using (PSTN, T1, etc). Speaking from a POTS line point of view, there can easily be a 7-10 second delay in the processing of DAHDI information (which would make your 1347 second call within tolerance). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart Sent: Thursday, November 03, 2011 5:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] duration limits in Dial() not being enforced at correct time Hi, We're trying to time-limit some calls by specifying L(x:y:z) as an option to the Dial command. If we set the limit to a fairly short duration (eg 120 seconds) then Asterisk seems to issue the hangup at about the right time. However, for longish calls we're seeing quite a bit of overspill. For example we tried to limit one to 1338 seconds but Asterisk didn't hang up until 1384 seconds after the call was answered. Also, the error is not always consistent - a second test call also limited to 1338 seconds was hung up by Asterisk after 1347 seconds. We saw this problem with Asterisk 1.6 but we've now tried on Asterisk 1.8.6.0 and are having the same problem. Here's a log from the Asterisk 1.8.6.0 box for the test call that should have been limited to 1338 seconds but was actually ended after 1384 seconds. The server wasn't carrying any other calls at the time or doing anything else so the load would have been very low. [Nov 2 16:47:37] VERBOSE[2029] pbx.c: -- Executing [01476292501@service_nts_v2:57] Dial(DAHDI/i2/7622323283-4, DAHDI/g1/08451238347,,L(1338000:3:5000)M(service-nts-v2-register-answer )) in new stack [Nov 2 16:47:37] VERBOSE[2029] features.c: Limit Data for this call: [Nov 2 16:47:37] VERBOSE[2029] features.c: timelimit = 1338000 ms (1338.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: play_warning = 3 ms (30.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: play_to_caller = yes [Nov 2 16:47:37] VERBOSE[2029] features.c: play_to_callee = no [Nov 2 16:47:37] VERBOSE[2029] features.c: warning_freq = 5000 ms (5.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: start_sound= [Nov 2 16:47:37] VERBOSE[2029] features.c: warning_sound = /var/lib/asterisk/sounds/bespoke/beep_200ms [Nov 2 16:47:37] VERBOSE[2029] features.c: end_sound = [Nov 2 16:47:37] VERBOSE[2029] sig_pri.c: -- Requested transfer capability: 0x00 - SPEECH [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- Called DAHDI/g1/08451238347 [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 is proceeding passing it to DAHDI/i2/7622323283-4 [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 is ringing [Nov 2 16:47:38] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 answered DAHDI/i2/7622323283-4 [Nov 2 16:47:38] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:1] NoOp(DAHDI/i1/08451238347-3, ANSWER MACRO) in new stack [Nov 2 16:47:38] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:2] AGI(DAHDI/i1/08451238347-3, agi://127.0.0.1:4573/ServiceNTSV2,mode=answered,uniqueID=1320252457.17_1,un iqueIDB=1320252457.18,ddi=08451238347,Goto=agiOK1) in new stack [Nov 2 16:47:39] VERBOSE[2029] res_agi.c: -- AGI Script Executing Application: (Goto) Options: (agiOK1) [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Goto (macro-service-nts-v2-register-answer,s,7) [Nov 2 16:47:39] VERBOSE[2029] res_agi.c: -- DAHDI/i1/08451238347-3AGI Script agi://127.0.0.1:4573/ServiceNTSV2 completed, returning 0 [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:7] GotoIf(DAHDI/i1/08451238347-3, 1?agiOK2) in new stack [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Goto (macro-service-nts-v2-register-answer,s,13) [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:13] NoOp(DAHDI/i1/08451238347-3, register-answer macro finished) in new stack [Nov 2
Re: [asterisk-users] duration limits in Dial() not being enforced at correct time
+1 Bryant - by using the Local/Context you are introducing some overhead to the process, but eliminating the dependence on DAHDI timing (not that there's anything wrong with that per se, but you can't control the Space Shuttle with a Bearcat Scanner (or can you?) ). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Thursday, November 03, 2011 8:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] duration limits in Dial() not being enforced at correct time If you dial to a Local/Context and use your time limits on that and then do your dial to your DAHDI device inside that context does that have any effect on the time limits working. We have used time limits with Local/Context dials and had them work with out any known issues. Thanks Bryant Zimmerman _ From: amit anand onewaytoconn...@gmail.com Sent: Thursday, November 03, 2011 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] duration limits in Dial() not being enforced at correct time On Thu, Nov 3, 2011 at 18:44, Danny Nicholas da...@debsinc.com wrote: Please elaborate on your flavor of DAHDI and LIBPRI and what type of DAHDI service you are using (PSTN, T1, etc). Speaking from a POTS line point of view, there can easily be a 7-10 second delay in the processing of DAHDI information (which would make your 1347 second call within tolerance). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart Sent: Thursday, November 03, 2011 5:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] duration limits in Dial() not being enforced at correct time Hi, We're trying to time-limit some calls by specifying L(x:y:z) as an option to the Dial command. If we set the limit to a fairly short duration (eg 120 seconds) then Asterisk seems to issue the hangup at about the right time. However, for longish calls we're seeing quite a bit of overspill. For example we tried to limit one to 1338 seconds but Asterisk didn't hang up until 1384 seconds after the call was answered. Also, the error is not always consistent - a second test call also limited to 1338 seconds was hung up by Asterisk after 1347 seconds. We saw this problem with Asterisk 1.6 but we've now tried on Asterisk 1.8.6.0 and are having the same problem. Here's a log from the Asterisk 1.8.6.0 box for the test call that should have been limited to 1338 seconds but was actually ended after 1384 seconds. The server wasn't carrying any other calls at the time or doing anything else so the load would have been very low. [Nov 2 16:47:37] VERBOSE[2029] pbx.c: -- Executing [01476292501@service_nts_v2:57] Dial(DAHDI/i2/7622323283-4, DAHDI/g1/08451238347,,L(1338000:3:5000)M(service-nts-v2-register-answer )) in new stack [Nov 2 16:47:37] VERBOSE[2029] features.c: Limit Data for this call: [Nov 2 16:47:37] VERBOSE[2029] features.c: timelimit = 1338000 ms (1338.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: play_warning = 3 ms (30.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: play_to_caller = yes [Nov 2 16:47:37] VERBOSE[2029] features.c: play_to_callee = no [Nov 2 16:47:37] VERBOSE[2029] features.c: warning_freq = 5000 ms (5.000 s) [Nov 2 16:47:37] VERBOSE[2029] features.c: start_sound= [Nov 2 16:47:37] VERBOSE[2029] features.c: warning_sound = /var/lib/asterisk/sounds/bespoke/beep_200ms [Nov 2 16:47:37] VERBOSE[2029] features.c: end_sound = [Nov 2 16:47:37] VERBOSE[2029] sig_pri.c: -- Requested transfer capability: 0x00 - SPEECH [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- Called DAHDI/g1/08451238347 [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 is proceeding passing it to DAHDI/i2/7622323283-4 [Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 is ringing [Nov 2 16:47:38] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 answered DAHDI/i2/7622323283-4 [Nov 2 16:47:38] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:1] NoOp(DAHDI/i1/08451238347-3, ANSWER MACRO) in new stack [Nov 2 16:47:38] VERBOSE[2029] pbx.c: -- Executing [s@macro-service-nts-v2-register-answer:2] AGI(DAHDI/i1/08451238347-3, agi://127.0.0.1:4573/ServiceNTSV2,mode=answered,uniqueID=1320252457.17_1,un http://127.0.0.1:4573/ServiceNTSV2,mode=answered,uniqueID=1320252457.17_1,u niqueIDB=1320252457.18,ddi=08451238347,Goto=agiOK1 iqueIDB=1320252457.18,ddi=08451238347,Goto=agiOK1) in new stack [Nov 2 16:47:39] VERBOSE[2029] res_agi.c: -- AGI Script Executing Application: (Goto) Options: (agiOK1) [Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Goto
Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks
2011/11/3 Danny Nicholas da...@debsinc.com What version of Asterisk? 1.6.1.18 Is the forwarding done using Followme, attended transfer or blind transfer? a plain Answer plus Dial ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Thursday, November 03, 2011 8:14 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks ** ** Hi, I'm still strugling with my CallerID presentation problem. Let me remind it : My setup is: Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob cellphone Ive configured Asterisk so that whenever Bob forwards its incoming call to its cellphone, the later phone should present Alice's number. I was originally told that sometimes Bob would be presented Alice's number, sometimes the dialed number (which in this case, also match the ANI or the receptionnist ID). Now, I can't certify this ever happened : maybe, it did happen, maybe not. What I can certify is this: 1. out of 32 different callers, 20 callers are presented with the correct number and 12 with the dialed number, 2. all those 32 callers are cellphones operated by the same (incumbent) telco which also operates ISDN, 3. Bob's cellphone is also operated by the same (incumbent) telco, 4. all this tries were done the same day, one after the other. The best explanation I can think of is this: Depending on the route used, the ANI is used instead of the presented caller ID. To prove that, I'll try to record 2 calls for the same caller and toward the same destination: one with the awaited presentation, one with a wrong one. (Sending this to the telco and have them change anything is an other story). Comments and suggestions are welcome. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks
Something like this?
[callbob]
Exten = start,1,answer
Exten = start,n,Dial(DAHDI/1/5551212,30)
If that is the case, Bob should always get the Caller ID of your asterisk
installation - I would suggest this instead
[callbob]
Exten = start,1,answer
Exten = start,n,Set(CALLERID(num)=${EXTEN})
Exten = start,n,Dial(DAHDI/1/5551212,30)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks
2011/11/3 Danny Nicholas da...@debsinc.com
What version of Asterisk?
1.6.1.18
Is the forwarding done using Followme, attended transfer or blind
transfer?
a plain Answer plus Dial
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks
Hi,
I'm still strugling with my CallerID presentation problem.
Let me remind it :
My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
cellphone
Ive configured Asterisk so that whenever Bob forwards its incoming call to
its cellphone, the later phone should present Alice's number.
I was originally told that sometimes Bob would be presented Alice's number,
sometimes the dialed number (which in this case, also match the ANI or the
receptionnist ID).
Now, I can't certify this ever happened : maybe, it did happen, maybe not.
What I can certify is this:
1. out of 32 different callers, 20 callers are presented with the correct
number and 12 with the dialed number,
2. all those 32 callers are cellphones operated by the same (incumbent)
telco which also operates ISDN,
3. Bob's cellphone is also operated by the same (incumbent) telco,
4. all this tries were done the same day, one after the other.
The best explanation I can think of is this:
Depending on the route used, the ANI is used instead of the presented
caller ID.
To prove that, I'll try to record 2 calls for the same caller and toward
the same destination: one with the awaited presentation, one with a wrong
one.
(Sending this to the telco and have them change anything is an other story).
Comments and suggestions are welcome.
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks
In your example the CallerID number will always be start. Not what he is
looking for.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, November 03, 2011 9:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks
Something like this?
[callbob]
Exten = start,1,answer
Exten = start,n,Dial(DAHDI/1/5551212,30)
If that is the case, Bob should always get the Caller ID of your asterisk
installation - I would suggest this instead
[callbob]
Exten = start,1,answer
Exten = start,n,Set(CALLERID(num)=${EXTEN})
Exten = start,n,Dial(DAHDI/1/5551212,30)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks
2011/11/3 Danny Nicholas da...@debsinc.com
What version of Asterisk?
1.6.1.18
Is the forwarding done using Followme, attended transfer or blind
transfer?
a plain Answer plus Dial
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks
Hi,
I'm still strugling with my CallerID presentation problem.
Let me remind it :
My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
cellphone
Ive configured Asterisk so that whenever Bob forwards its incoming call
to its cellphone, the later phone should present Alice's number.
I was originally told that sometimes Bob would be presented Alice's
number, sometimes the dialed number (which in this case, also match the ANI or
the receptionnist ID).
Now, I can't certify this ever happened : maybe, it did happen, maybe
not.
What I can certify is this:
1. out of 32 different callers, 20 callers are presented with the
correct number and 12 with the dialed number,
2. all those 32 callers are cellphones operated by the same (incumbent)
telco which also operates ISDN,
3. Bob's cellphone is also operated by the same (incumbent) telco,
4. all this tries were done the same day, one after the other.
The best explanation I can think of is this:
Depending on the route used, the ANI is used instead of the presented
caller ID.
To prove that, I'll try to record 2 calls for the same caller and
toward the same destination: one with the awaited presentation, one with a
wrong one.
(Sending this to the telco and have them change anything is an other
story).
Comments and suggestions are welcome.
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID inconsistently presented through ISDN/cellular networks
Trying to save a few keystrokes - better example
[callbob]
Exten = _XX.,1,answer
Exten = _XX.,n,Dial(DAHDI/1/5551212,30)
If that is the case, Bob should always get the Caller ID of your asterisk
installation - I would suggest this instead
[callbob]
Exten = _XX.,1,answer
Exten = _XX.,n,Set(CALLERID(num)=${EXTEN})
Exten = _XX.,n,Dial(DAHDI/1/5551212,30)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Thursday, November 03, 2011 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks
In your example the CallerID number will always be start. Not what he is
looking for.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, November 03, 2011 9:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks
Something like this?
[callbob]
Exten = start,1,answer
Exten = start,n,Dial(DAHDI/1/5551212,30)
If that is the case, Bob should always get the Caller ID of your asterisk
installation - I would suggest this instead
[callbob]
Exten = start,1,answer
Exten = start,n,Set(CALLERID(num)=${EXTEN})
Exten = start,n,Dial(DAHDI/1/5551212,30)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks
2011/11/3 Danny Nicholas da...@debsinc.com
What version of Asterisk?
1.6.1.18
Is the forwarding done using Followme, attended transfer or blind
transfer?
a plain Answer plus Dial
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented
through ISDN/cellular networks
Hi,
I'm still strugling with my CallerID presentation problem.
Let me remind it :
My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM--
Bob cellphone
Ive configured Asterisk so that whenever Bob forwards its incoming
call to its cellphone, the later phone should present Alice's number.
I was originally told that sometimes Bob would be presented Alice's
number, sometimes the dialed number (which in this case, also match the ANI
or the receptionnist ID).
Now, I can't certify this ever happened : maybe, it did happen,
maybe not.
What I can certify is this:
1. out of 32 different callers, 20 callers are presented with the
correct number and 12 with the dialed number,
2. all those 32 callers are cellphones operated by the same
(incumbent) telco which also operates ISDN,
3. Bob's cellphone is also operated by the same (incumbent) telco,
4. all this tries were done the same day, one after the other.
The best explanation I can think of is this:
Depending on the route used, the ANI is used instead of the
presented caller ID.
To prove that, I'll try to record 2 calls for the same caller and
toward the same destination: one with the awaited presentation, one with a
wrong one.
(Sending this to the telco and have them change anything is an other
story).
Comments and suggestions are welcome.
Regards
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Re: [asterisk-users] bug in queuemanager?
Anyone?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: dinsdag 1 november 2011 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?
Sorry it took me a while, but I was ill for a few days J
Part1: http://pastebin.com/SZqgxh7B
Part2: http://pastebin.com/gfJtVVRE
In this log a call from extension 346 is made to queue 900.
Queue 900 has 1 agent namely agent 300 which is logged on at extension
204.
Queue 901 has 1 agent namely agent 301 which is logged on at extension
203.
Agent 300 answers call from 346 and transfers this call to queue 901.
After agent 301 has answered this forwarded call (caller 346) a new call
from 346 arrives at queue 901.
After agent 301 hangs up the call, the new call from 346 is presented
immediately without any wrap-up time.
Hope this logging helps...
Greetings,
Henry
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren
Selby
Sent: dinsdag 25 oktober 2011 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?
On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger h.dog...@telecats.nl
wrote:
Customer 200 calls to queue 900, Agent 300 answers but tells Customer
200 that he should be at Queue 901 and transfers Customer 200 (using *2)
to Queue 901. Agent 301 now gets the call from Queue 901 with Customer
200, answers the calls etc. After disconnect a new call arrivers
immediately from Queue 901, without any wrap-up time. This should be
considered as a bug IMO.
Any ideas on how to fix, workaround this problem?
Please share the CLI output of such a situation, with the verbosity and
debugging both set to 10 ('core set verbose 10' and 'core set debug 10'
from the asterisk CLI), it may shed some light on whether this is a bug
or a feature.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com
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[asterisk-users] Asterisk as SoftSwitch - Hardware
Hi list, Could anyone tell me what is the recommended hardware to a system for following configuration: SBC -- Asterisk (SS) -- Carrier GW Asterisk should work as a Class 4 SoftSwitch, with following functionalists: - Do the IP Authentication - All communications on RTP/G729 (no transcoding required) - Load of 1200 concurrent call sessions - No call routing required Thanks in advance, Sunny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as SoftSwitch - Hardware
Shouldn't you be using a Proxy? Nick. On Thu, Nov 3, 2011 at 1:04 PM, Sunny no7f...@gmail.com wrote: Hi list, Could anyone tell me what is the recommended hardware to a system for following configuration: SBC -- Asterisk (SS) -- Carrier GW Asterisk should work as a Class 4 SoftSwitch, with following functionalists: - Do the IP Authentication - All communications on RTP/G729 (no transcoding required) - Load of 1200 concurrent call sessions - No call routing required Thanks in advance, Sunny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as SoftSwitch - Hardware
I was thinking in Kamailio, but this sip proxy handles only the SIP signalling traffic, no media processing. On 3 November 2011 17:07, Nick Khamis sym...@gmail.com wrote: Shouldn't you be using a Proxy? Nick. On Thu, Nov 3, 2011 at 1:04 PM, Sunny no7f...@gmail.com wrote: Hi list, Could anyone tell me what is the recommended hardware to a system for following configuration: SBC -- Asterisk (SS) -- Carrier GW Asterisk should work as a Class 4 SoftSwitch, with following functionalists: - Do the IP Authentication - All communications on RTP/G729 (no transcoding required) - Load of 1200 concurrent call sessions - No call routing required Thanks in advance, Sunny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as SoftSwitch - Hardware
Sunny- I was thinking in Kamailio, but this sip proxy handles only the SIP signalling traffic, no media processing. Kamailio + rtpproxy. -Jeff On 3 November 2011 17:07, Nick Khamis sym...@gmail.com wrote: Shouldn't you be using a Proxy? Nick. On Thu, Nov 3, 2011 at 1:04 PM, Sunny no7f...@gmail.com wrote: Hi list, Could anyone tell me what is the recommended hardware to a system for following configuration: SBC -- Asterisk (SS) -- Carrier GW Asterisk should work as a Class 4 SoftSwitch, with following functionalists: - Do the IP Authentication - All communications on RTP/G729 (no transcoding required) - Load of 1200 concurrent call sessions - No call routing required Thanks in advance, Sunny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2 pbxes
if i run let's say 1 pbx running on my main linux box and a another on my windows box if a person dial my main number and press lets say 1 are it possible to transfer the call over to my other pbx hope anyone understand-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 pbxes
Yes. If you have two asterisk boxes running you can trunk them together and place calls from one to to the other. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 3, 2011, at 11:36 AM, mattias wrote: if i run let's say 1 pbx running on my main linux box and a another on my windows box if a person dial my main number and press lets say 1 are it possible to transfer the call over to my other pbx hope anyone understand -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 pbxes
ok so if i have a automatic phonesystem on the first i can e.g press 1 forpersonal service sounds cool - Original Message - From: Jim Dickenson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 03, 2011 8:40 PM Subject: Re: [asterisk-users] 2 pbxes Yes. If you have two asterisk boxes running you can trunk them together and place calls from one to to the other. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 3, 2011, at 11:36 AM, mattias wrote: if i run let's say 1 pbx running on my main linux box and a another on my windows box if a person dial my main number and press lets say 1 are it possible to transfer the call over to my other pbx hope anyone understand -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DID from Direct from Telco
Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can paint a picture on how the DID suppliers function it would be greatly appreciated. If I were to guess it would be: Telco Lines - Gateway E1/T1 - SIP Proxy - Media Servers? With this scenario, do we now have control over the number of channels? Thanks in Advance, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] any live queue monitor recommendation?
Hi asterisk users, can any recommend me a live queue monitor for asterisk queues? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue status question.
Hi all, can any tell me why queue status don't shows the statistics or number of calls with exitwithkey status? it only shows the number of waiting, completed and abandoned calls in a queue. Regards, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID from Direct from Telco
On 11/03/2011 07:20 PM, Nick Khamis wrote: Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can paint a picture on how the DID suppliers function it would be greatly appreciated. If I were to guess it would be: Telco Lines - Gateway E1/T1 - SIP Proxy - Media Servers? With this scenario, do we now have control over the number of channels? Thanks in Advance, Simplest (with 3-4 T1s): Telco Lines - Asterisk box with T1 card (and possibly a codec processor card) - Customer More complex (with a bunch of circuits) : Telco Lines - Gateway T1 - SIP Proxy - Media Servers - Customer And if your question of number of channels is Can I control the number of channels a customer can use simultaneously?, then the answer is With Asterisk, Yes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID from Direct from Telco
Hello James, Thank you so much for your response. We just purchased an AudioCodes MP124 for this job. And setting up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the Telco here in Toronto. As for other Telcos around the world, for example Bell South in the states, is it possible to have them route a block of Florida phone numbers to our FXS port here in Canada, or do we have to have a T1 gateway + SIP Proxy in Florida, routing the calls to our setup in Toronto and vice versa? Thanks in Advance, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID from Direct from Telco
On 11/03/2011 09:16 PM, Nick Khamis wrote: Hello James, Thank you so much for your response. We just purchased an AudioCodes MP124 for this job. And setting up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the Telco here in Toronto. As for other Telcos around the world, for example Bell South in the states, is it possible to have them route a block of Florida phone numbers to our FXS port here in Canada, or do we have to have a T1 gateway + SIP Proxy in Florida, routing the calls to our setup in Toronto and vice versa? Routing Florida numbers up to Canada would get you charged LD per minute fees. You can go with a provider like Level 3 or Global Crossing and they can hand you a T1 circuit that has DIDs from many different areas in the US. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID from Direct from Telco
Fair enough, In regards to the the diagram discussed earlier: Telco Lines - Gateway T1 - SIP Proxy - Media Servers - Customer I understand that a T1 Gateway that has 480 channels, can handle up to 240 calls. That is more than enough for the Gateway T1 - SIP Proxy part of the diagram. I just want to make terribly sure I understand the Telco Lines - Gateway T1. If the Gateway T1 plugs into only 1 FXS port, is that FXS port only capable of handling 2 channels, i.e., one call? Thanks in Advnace, Nick. On Thu, Nov 3, 2011 at 9:24 PM, James Sharp ja...@fivecats.org wrote: On 11/03/2011 09:16 PM, Nick Khamis wrote: Hello James, Thank you so much for your response. We just purchased an AudioCodes MP124 for this job. And setting up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the Telco here in Toronto. As for other Telcos around the world, for example Bell South in the states, is it possible to have them route a block of Florida phone numbers to our FXS port here in Canada, or do we have to have a T1 gateway + SIP Proxy in Florida, routing the calls to our setup in Toronto and vice versa? Routing Florida numbers up to Canada would get you charged LD per minute fees. You can go with a provider like Level 3 or Global Crossing and they can hand you a T1 circuit that has DIDs from many different areas in the US. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] custom automated meeting
on 10/31/2011 11:59 PM Thanasis wrote the following:
I need your help in implementing the following scenario:
A certain extension will ring two sip phones simultaneously and when one
of them answers, the other keeps ringing until it answers too, and then
all three (the caller and the other two) are immediately placed in a
conference room (same room for all three).
Can we do it?
FWIW, using call files:
Here is the relevant section of the dialplan:
exten = 300,1,Noop(creating conference)
same = n,Set(conf_name=conf-${RAND(1,1000)})
same = n,System(/etc/asterisk/scripts/callgenerator SIP/dev1
${conf_name})
same = n,System(/etc/asterisk/scripts/callgenerator SIP/dev2
${conf_name})
same = n,MeetMe(${conf_name},dFI1xAC)
same = n,Noop(do post conference stuff)
... and here is the script /etc/asterisk/scripts/callgenerator:
#!/bin/bash
PHONE=$(echo $1 |cut -f2 -d/)
ROOM=$2
echo Channel: $1 /var/spool/asterisk/tmp/${PHONE}.call
echo Application: MeetMe /var/spool/asterisk/tmp/${PHONE}.call
echo Data: ${ROOM},dFI1x /var/spool/asterisk/tmp/${PHONE}.call
mv -f /var/spool/asterisk/tmp/${PHONE}.call /var/spool/asterisk/outgoing
PS: Thanks much to Yaroslav for his help :)
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Re: [asterisk-users] DID from Direct from Telco
The mp124 is a analog gateway and doesn't support t1's I think A T1 is a digital line which has 24 channels per port which means 24 calls concurrently if you want more channels you need more ports DID's are incoming numbers the telco sends down your trunk(port) you could have thousands of DID's on 1 T1 You need a digital gateway for connecting to a T1 Did you check if your provider will give you a T1 or maybe they could provide you a sip trunk which will save you on the hardware -Original Message- From: Nick Khamis sym...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 3 Nov 2011 22:10:31 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DID from Direct from Telco Fair enough, In regards to the the diagram discussed earlier: Telco Lines - Gateway T1 - SIP Proxy - Media Servers - Customer I understand that a T1 Gateway that has 480 channels, can handle up to 240 calls. That is more than enough for the Gateway T1 - SIP Proxy part of the diagram. I just want to make terribly sure I understand the Telco Lines - Gateway T1. If the Gateway T1 plugs into only 1 FXS port, is that FXS port only capable of handling 2 channels, i.e., one call? Thanks in Advnace, Nick. On Thu, Nov 3, 2011 at 9:24 PM, James Sharp ja...@fivecats.org wrote: On 11/03/2011 09:16 PM, Nick Khamis wrote: Hello James, Thank you so much for your response. We just purchased an AudioCodes MP124 for this job. And setting up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the Telco here in Toronto. As for other Telcos around the world, for example Bell South in the states, is it possible to have them route a block of Florida phone numbers to our FXS port here in Canada, or do we have to have a T1 gateway + SIP Proxy in Florida, routing the calls to our setup in Toronto and vice versa? Routing Florida numbers up to Canada would get you charged LD per minute fees. You can go with a provider like Level 3 or Global Crossing and they can hand you a T1 circuit that has DIDs from many different areas in the US. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID from Direct from Telco
One FXS port can only handle one call. A PRI T1 gateway can handle 23 call channels. A single T1 Data line with SIP can handle about 18 call channels running G711, 37 channels running g729 Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Nick Khamis sym...@gmail.com Sent: Thursday, November 03, 2011 10:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DID from Direct from Telco Fair enough, In regards to the the diagram discussed earlier: Telco Lines - Gateway T1 - SIP Proxy - Media Servers - Customer I understand that a T1 Gateway that has 480 channels, can handle up to 240 calls. That is more than enough for the Gateway T1 - SIP Proxy part of the diagram. I just want to make terribly sure I understand the Telco Lines - Gateway T1. If the Gateway T1 plugs into only 1 FXS port, is that FXS port only capable of handling 2 channels, i.e., one call? Thanks in Advnace, Nick. On Thu, Nov 3, 2011 at 9:24 PM, James Sharp ja...@fivecats.org wrote: On 11/03/2011 09:16 PM, Nick Khamis wrote: Hello James, Thank you so much for your response. We just purchased an AudioCodes MP124 for this job. And setting up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the Telco here in Toronto. As for other Telcos around the world, for example Bell South in the states, is it possible to have them route a block of Florida phone numbers to our FXS port here in Canada, or do we have to have a T1 gateway + SIP Proxy in Florida, routing the calls to our setup in Toronto and vice versa? Routing Florida numbers up to Canada would get you charged LD per minute fees. You can go with a provider like Level 3 or Global Crossing and they can hand you a T1 circuit that has DIDs from many different areas in the US. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID from Direct from Telco
Thanks bryant ur right 23 channels I'm used to E1's where you a get 30 channels a even number -Original Message- From: Bryant Zimmerman brya...@zktech.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 3 Nov 2011 22:32:41 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: brya...@zktech.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DID from Direct from Telco -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
