Re: [asterisk-users] Exiting the queue doesn't work
On 3/4/2013 6:27 AM, Gertjan Baarda wrote: Dear guru's Hopefully someone can shed some light in my issue. I have created a queue with a ringall strategy and all works fine. I want a caller to be able to exit the queue so they can leave a message. I've added the H option in queue command so callers can press * to exit. So far all well, on the cli there is a message the caller pressed * and the extensions stops ringing. But here's the thing: the caller stays in the queue and after a few seconds the extensions starts to ring again. I want the call to leave the queue and continue in the dialplan. After extensively googling the issue, I've found everything (also bug related), accept my answer. What am I missing here? It's Asterisk 1.8 on a Debianbox. Thanks! Gertjan -- Look at context= in queues.conf. Alex -- _ -- 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] Exiting the queue doesn't work
On 3/4/2013 7:27 AM, Gertjan Baarda wrote: ok, resumé: When I use the n option in the queue command I can let the caller exit the queue and send the call to a IVR-ish context and ask if he wants to leave a message. I can timeout this an then place the call back in the queue. When I use this approach, what will the new position be of the caller? Not back in line I hope? On Mon, Mar 4, 2013 at 2:12 PM, Bharat Lalcheta bharatlalch...@gmail.com mailto:bharatlalch...@gmail.com wrote: yes, context parameter in queue.conf is more likely option for you. It will work during MOH too. On Mon, Mar 4, 2013 at 6:33 PM, Bharat Lalcheta bharatlalch...@gmail.com mailto:bharatlalch...@gmail.com wrote: No its again place into queue so its start with new available position. However, mostly all users remain in same position if he come again in queue using below scenario. Regards, Bharat Lalcheta On Mon, Mar 4, 2013 at 6:22 PM, Gertjan Baarda gertjan.baa...@gmail.com mailto:gertjan.baa...@gmail.com wrote: Ah.. thanks! That was the light I needed. When the caller is placed back in the queue, I presume the caller remain it's position in the queue? On Mon, Mar 4, 2013 at 1:45 PM, Bharat Lalcheta bharatlalch...@gmail.com mailto:bharatlalch...@gmail.com wrote: Hii, Queue(testq,H) feature works once call connected with agent i.e. not work during MOH. Also once you disconnect call using H (*) option, it will not useful to leave voicemail. Instead you use, queue timeout option and ask caller to leave voice mail if he wants else put back him to queue again. Hope it helps you out. Regards, Bharat Lalcheta On Mon, Mar 4, 2013 at 5:57 PM, Gertjan Baarda gertjan.baa...@gmail.com mailto:gertjan.baa...@gmail.com wrote: Dear guru's Hopefully someone can shed some light in my issue. I have created a queue with a ringall strategy and all works fine. I want a caller to be able to exit the queue so they can leave a message. I've added the H option in queue command so callers can press * to exit. So far all well, on the cli there is a message the caller pressed * and the extensions stops ringing. But here's the thing: the caller stays in the queue and after a few seconds the extensions starts to ring again. I want the call to leave the queue and continue in the dialplan. After extensively googling the issue, I've found everything (also bug related), accept my answer. What am I missing here? It's Asterisk 1.8 on a Debianbox. Thanks! Gertjan -- _ -- 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 -- Bharat Lalcheta -- _ -- 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 -- Bharat Lalcheta -- Bharat Lalcheta Why not set context= some context and use periodic announcement to say You may press * at any time to leave us a message. You can also play the message before entering the queue (only once and caller may forget what key to press). This way the caller looses their position in the queue only if they choose to leave a message. Alex -- _ -- 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
Re: [asterisk-users] Exiting the queue doesn't work
On 3/4/2013 8:00 AM, Gertjan Baarda wrote: This will only work with the n option in the queue command and retry=0 in queue.conf. Is it not? On Mon, Mar 4, 2013 at 2:55 PM, Alex Kauffmann akauf...@prodigy.net.mx mailto:akauf...@prodigy.net.mx wrote: On 3/4/2013 7:27 AM, Gertjan Baarda wrote: ok, resumé: When I use the n option in the queue command I can let the caller exit the queue and send the call to a IVR-ish context and ask if he wants to leave a message. I can timeout this an then place the call back in the queue. When I use this approach, what will the new position be of the caller? Not back in line I hope? On Mon, Mar 4, 2013 at 2:12 PM, Bharat Lalcheta bharatlalch...@gmail.com mailto:bharatlalch...@gmail.com mailto:bharatlalcheta@gmail.__com mailto:bharatlalch...@gmail.com wrote: yes, context parameter in queue.conf is more likely option for you. It will work during MOH too. On Mon, Mar 4, 2013 at 6:33 PM, Bharat Lalcheta bharatlalch...@gmail.com mailto:bharatlalch...@gmail.com mailto:bharatlalcheta@gmail.__com mailto:bharatlalch...@gmail.com wrote: No its again place into queue so its start with new available position. However, mostly all users remain in same position if he come again in queue using below scenario. Regards, Bharat Lalcheta On Mon, Mar 4, 2013 at 6:22 PM, Gertjan Baarda gertjan.baa...@gmail.com mailto:gertjan.baa...@gmail.com mailto:gertjan.baarda@gmail.__com mailto:gertjan.baa...@gmail.com wrote: Ah.. thanks! That was the light I needed. When the caller is placed back in the queue, I presume the caller remain it's position in the queue? On Mon, Mar 4, 2013 at 1:45 PM, Bharat Lalcheta bharatlalch...@gmail.com mailto:bharatlalch...@gmail.com mailto:bharatlalcheta@gmail.__com mailto:bharatlalch...@gmail.com wrote: Hii, Queue(testq,H) feature works once call connected with agent i.e. not work during MOH. Also once you disconnect call using H (*) option, it will not useful to leave voicemail. Instead you use, queue timeout option and ask caller to leave voice mail if he wants else put back him to queue again. Hope it helps you out. Regards, Bharat Lalcheta On Mon, Mar 4, 2013 at 5:57 PM, Gertjan Baarda gertjan.baa...@gmail.com mailto:gertjan.baa...@gmail.com mailto:gertjan.baarda@gmail.__com mailto:gertjan.baa...@gmail.com wrote: Dear guru's Hopefully someone can shed some light in my issue. I have created a queue with a ringall strategy and all works fine. I want a caller to be able to exit the queue so they can leave a message. I've added the H option in queue command so callers can press * to exit. So far all well, on the cli there is a message the caller pressed * and the extensions stops ringing. But here's the thing: the caller stays in the queue and after a few seconds the extensions starts to ring again. I want the call to leave the queue and continue in the dialplan. After extensively googling the issue, I've found everything (also bug related), accept my answer. What am I missing here? It's Asterisk 1.8 on a Debianbox. Thanks! Gertjan -- _ -- 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
Re: [asterisk-users] PRI can receive calls but cannot dial out
On 12/7/2012 6:23 AM, Vieri wrote: Am 05.12.2012 08:48, schrieb Vieri: Hi, I'm trying to call out from a SIP extension to an outbound destination via a PRI E1 (Digium B410P). Please take a look at the PRI debug below. # cat /etc/dahdi/system.conf # Digium Wildcard TDM400P REV I (WCTDM/4) fxsks=1 echocanceller=oslec,1 fxsks=2 echocanceller=oslec,2 fxsks=3 echocanceller=oslec,3 fxsks=4 echocanceller=oslec,4 # Digium Wildcard TDM2400P (WCTDM/0) fxsks=5 echocanceller=oslec,5 fxsks=6 echocanceller=oslec,6 fxsks=7 echocanceller=oslec,7 fxsks=8 echocanceller=oslec,8 fxsks=9 echocanceller=oslec,9 fxsks=10 echocanceller=oslec,10 fxsks=11 echocanceller=oslec,11 fxsks=12 echocanceller=oslec,12 # Digium Wildcard B410P (B4/0/1) span=3,1,0,CCS,AMI bchan=29-30 hardhdlc=31 echocanceller=oslec,29-30 # Digium Wildcard B410P (B4/0/2) span=4,2,0,CCS,AMI bchan=32-33 hardhdlc=34 echocanceller=oslec,32-33 # Digium Wildcard B410P (B4/0/3) span=5,3,0,CCS,AMI bchan=35-36 hardhdlc=37 echocanceller=oslec,35-36 # Digium Wildcard B410P (B4/0/4) span=6,4,0,CCS,AMI bchan=38-39 hardhdlc=40 echocanceller=oslec,38-39 # lsmod | grep wcb4xxp wcb4xxp 66250 12 dahdi 169899 65 dahdi_echocan_oslec,wcb4xxp,wctdm24xxp,dahdi_voicebus,wctdm # cat chan_dahdi.conf [trunkgroups] [channels] transfer = yes usecallerid = yes cidsignalling = dtmf callwaiting = yes usecallingpres = yes callwaitingcallerid = yes threewaycalling = yes canpark = yes cancallforward = yes callreturn = yes callprogress = no overlapdial = yes echocancel = yes facilityenable = yes immediate = no busydetect = no ; Digium Wildcard TDM400P REV I (WCTDM/4) signalling = fxs_ks txgain = 1.0 rxgain = 14.0 group = 3 context = incoming-dahdi-3 faxdetect = incoming channel = 1,2,3,4 ; Digium Wildcard TDM2400P (WCTDM/0) group = 4 context = incoming-dahdi-4 faxdetect = incoming channel = 5,6,7,8,9,10,11,12 ; Digium Wildcard B410P (B4/0/1) signalling = bri_cpe switchtype = euroisdn rxgain = 2.0 group = 2 context = incoming-dahdi-2 faxdetect = incoming channel = 29-30 ; Digium Wildcard B410P (B4/0/2) channel = 32-33 ; Digium Wildcard B410P (B4/0/3) channel = 35-36 ; Digium Wildcard B410P (B4/0/4) channel = 38-39 --- # asterisk -rx dahdi show status Description Alarms IRQbpviol CRC Fra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) Wildcard TDM2400P OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 1 RED 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 2 OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 3 OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) B4XXP (PCI) Card 0 Span 4 OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) Note that I have 3 cables connected and 1 port is free (RED). --- in AEL dialplan, I run: Dial(DAHDI/g2/XX); in the *CLI I see the following: -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/g2/XX -- Span 4: Channel 0/1 got hangup, cause 18 -- Hungup 'DAHDI/i4/XX-7' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/4053-0089' status is 'CHANUNAVAIL' If I enable PRI debug: -- Executing [@company:1] Dial(SIP/4053-0001, DAHDI/g2/XX) in new stack PRI Span: 4 -- Making new call for cref 32772 -- Requested transfer capability: 0x00 - SPEECH PRI Span: 4 PRI Span: 4 DL-DATA request PRI Span: 4 Protocol Discriminator: Q.931 (8) len=32 PRI Span: 4 TEI=0 Call Ref: len= 1 (reference 4/0x4) (Sent from originator) PRI Span: 4 Message Type: SETUP (5) PRI Span: 4 TEI=0 Transmitting N(S)=6, window is open V(A)=6 K=1 PRI Span: 4 PRI Span: 4 Protocol Discriminator: Q.931 (8) len=32 PRI Span: 4 TEI=0 Call Ref: len= 1 (reference 4/0x4) (Sent from originator) PRI Span: 4 Message Type: SETUP (5) PRI Span: 4 [04 03 80 90 a3] PRI Span: 4 Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) PRI Span: 4 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) PRI Span: 4 User information layer 1: A-Law (35) PRI Span: 4 [18 01 81] PRI Span: 4 Channel ID (len= 3) [ Ext: 1 IntID: Implicit BRI Spare: 0 Preferred Dchan: 0 PRI Span: 4 ChanSel: B1 channel PRI Span: 4 ] PRI Span: 4 [6c 06 21 80 34 30 35 33] PRI Span: 4 Calling Party Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) PRI Span: 4 Presentation: Presentation allowed, User-provided, not screened (0) '4053' ] PRI Span: 4 [70 0a 80 36 35 36 36 36 30 34 39 39] PRI Span: 4 Called Party Number (len=12) [
Re: [asterisk-users] Change phone display from queue calls
On 12/6/2012 12:32 PM, Carlos Alvarez wrote: We are trying to set up a system where the calls from the queue show a specific name or number on the phone. The calls would come into one of a few dozen DID numbers, each one for a specific company. The agent needs to know which company the call is for and answer appropriately. I've done a lot of this in dialplans but haven't found a way to do it in a queue. -- Carlos Alvarez TelEvolve 602-889-3003 We either set callerid(name) with a fixed name depending on the trunk, or set it from a variable we get from a database based on callerid(number) before we dial the queue. The new value for callerid(name) will show on the agent's screen. Setting Callerid(number) will work as well. alex -- _ -- 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] leading ghost 0
On 11/21/2012 10:53 AM, gincantalupo wrote: Alex, I had already tried itreloading chan_dahdi.so module is enough...I saw Asterisk was behaving differently after reload. To tell the truth, setting pridialplan=unknown causes Asterisk to stop reading following channels configuration...it says pridialplan is already unknown so it stops evaluating chan_dahdi.conf file useless to say that all n+1 channels do not work. Maybe it is a bug but with that parameter set in that way I cannot dial. I'm sure Asterisk is dialling the right number: [2012-11-21 09:05:29] VERBOSE[8314] logger.c: [70 0b a1 33 34 39 3x 3x 3x 3x 3x 3x 34] [2012-11-21 09:05:29] VERBOSE[8314] logger.c: Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '3497078884' ] [2012-11-21 09:05:29] VERBOSE[8314] logger.c: q931.c:3134 q931_setup: call 32781 on channel 6 enters state 1 (Call Initiated) [2012-11-21 09:05:29] VERBOSE[8314] logger.c: -- Called 6/349xx4 I'm starting to think it is a telco problem... in case I'd change some parameter like pridialplan or similar, shouldn't I just see a leading 0 in the frame like this: [70 0b a1 *30* 33 34 39 3x 3x 3x 3x 3x 3x 34] added by Asterisk/DAHDI?? I've used this page as reference about frame fields: http://www.acacia-net.com/wwwcla/protocol/q931_ie.htm Thank you. Giorgio Incantalupo On 11/20/2012 05:23 PM, Alex Kauffmann wrote: On 11/20/2012 8:03 AM, gincantalupo wrote: Hi Leandro, I'm sure nobody has added something... tried prilocaldialplan and pridialplan but nothing changed. Question: if pridialplan or prilocaldialplan would work, should I see the 0 inside PRI frame with intense debug or it is hidden? Yes...the technician did it...there is only one cable. Maybe it is the socket circuitry that has something wrong but I do not know ho to check. Asap I'll be on site I'll do more testing. Thank you Giorgio On 11/20/2012 01:13 PM, Leandro Dardini wrote: That is a real mistery! I like a lots these cases when all seems not working despite all being correctly configured, but you know first or later you'll find the answer. From your website, it seems you are selling/renting PBX based on asterisk, so you can be sure nobody has messed with the asterisk or dahdi source code adding a zero... I am sure you have already tried with a brand new server. Have you checked the pridialplan and prilocaldialplan setting? If I was in your shoes, I'll get another server, with a PRI configured as master and hook it at your PBX to really check if the zero is sent. Does the technician try to make phone calls from the same network cable you are using? Leandro 2012/11/20 gincantalupo gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com Hi Leandro, thanks for your answer. I already have tried those parameters but without any positive result. The telco technician has tried the line with its machine and it worked...remote telco technicians say they get a leading zero... I'm thinking there is something strange in the middle that adds the zero but do not know what it is. Strange is the fact that you can call some numbers with or without the prefix zero... Moreover we had no problem with the previous telco (fastweb). So we can only call PTSN numbersnot mobile phones. Giorgio On 11/20/2012 11:12 AM, Leandro Dardini wrote: 2012/11/20 gincantalupo gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com Hi all, I have problems dialling out because my new telco (the previous gave no problems) tells me my PBX adds a leading 0 and that's why I cannot dial out (but I can receive calls). I make a small extensions.conf as a test: exten = 666,1,Dial(DAHDI/g1/339xx) but cannot dial out Curious thing is that exten = 666,1,Dial(DAHDI/g1/0233xx) and exten = 666,1,Dial(DAHDI/g1/233xx) call the same number!!! Line in use is a PRI. My Asterisk version is 1.4.26.2 dahdi version: 2.2.0.2 wanpipe-3.4.6 I checked with intense pri debug and see no 0 inside frames How can I really be SURE Asterisk is not adding some leading zero? Thank you. Giorgio. I have never heard of a way to automatically add digits when using PRI, however can you check your chan_dahdi.conf about the following lines: internationalprefix = nationalprefix = localprefix = If presents, try messing with them. If you are using the PRI in Italy, every provider has PRI configured in its own way, some time even the same provider is configuring PRI lines in multiple times, but often the problems are on receiving the calls (like calls with and without the area code, with or without the leading zero, etc. etc.) Leandro -- The prilocaldialplan parameter
Re: [asterisk-users] leading ghost 0
On 11/20/2012 8:03 AM, gincantalupo wrote: Hi Leandro, I'm sure nobody has added something... tried prilocaldialplan and pridialplan but nothing changed. Question: if pridialplan or prilocaldialplan would work, should I see the 0 inside PRI frame with intense debug or it is hidden? Yes...the technician did it...there is only one cable. Maybe it is the socket circuitry that has something wrong but I do not know ho to check. Asap I'll be on site I'll do more testing. Thank you Giorgio On 11/20/2012 01:13 PM, Leandro Dardini wrote: That is a real mistery! I like a lots these cases when all seems not working despite all being correctly configured, but you know first or later you'll find the answer. From your website, it seems you are selling/renting PBX based on asterisk, so you can be sure nobody has messed with the asterisk or dahdi source code adding a zero... I am sure you have already tried with a brand new server. Have you checked the pridialplan and prilocaldialplan setting? If I was in your shoes, I'll get another server, with a PRI configured as master and hook it at your PBX to really check if the zero is sent. Does the technician try to make phone calls from the same network cable you are using? Leandro 2012/11/20 gincantalupo gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com Hi Leandro, thanks for your answer. I already have tried those parameters but without any positive result. The telco technician has tried the line with its machine and it worked...remote telco technicians say they get a leading zero... I'm thinking there is something strange in the middle that adds the zero but do not know what it is. Strange is the fact that you can call some numbers with or without the prefix zero... Moreover we had no problem with the previous telco (fastweb). So we can only call PTSN numbersnot mobile phones. Giorgio On 11/20/2012 11:12 AM, Leandro Dardini wrote: 2012/11/20 gincantalupo gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com Hi all, I have problems dialling out because my new telco (the previous gave no problems) tells me my PBX adds a leading 0 and that's why I cannot dial out (but I can receive calls). I make a small extensions.conf as a test: exten = 666,1,Dial(DAHDI/g1/339xx) but cannot dial out Curious thing is that exten = 666,1,Dial(DAHDI/g1/0233xx) and exten = 666,1,Dial(DAHDI/g1/233xx) call the same number!!! Line in use is a PRI. My Asterisk version is 1.4.26.2 dahdi version: 2.2.0.2 wanpipe-3.4.6 I checked with intense pri debug and see no 0 inside frames How can I really be SURE Asterisk is not adding some leading zero? Thank you. Giorgio. I have never heard of a way to automatically add digits when using PRI, however can you check your chan_dahdi.conf about the following lines: internationalprefix = nationalprefix = localprefix = If presents, try messing with them. If you are using the PRI in Italy, every provider has PRI configured in its own way, some time even the same provider is configuring PRI lines in multiple times, but often the problems are on receiving the calls (like calls with and without the area code, with or without the leading zero, etc. etc.) Leandro -- The prilocaldialplan parameter is for inbound so you would have seen no changes. Did you try: pridialplan=unknown Did you restart dahdi and asterisk after the changes? Alex -- _ -- 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] How to tie orders taken to specific CDR records
On 10/25/2012 11:18 AM, Mitch Claborn wrote: Our phone operators work off of an Asterisk queue. They take calls from customers and take orders with our back end systems. What I need to be able to do is tie the orders taken to the specific CDR record that reflects the call from which the order originated. The typical/sample CDR table doesn't have a primary key. I can add an auto-generated PK, but the CDR is not written until the call ends, when the orders have already been placed. (Even if the CDR was written earlier, could I retrieve the generated PK from it in the dialplan somehow?) Is there some combination of fields in the CDR that might uniquely identify a specific call? Open to any and all ideas. Try looking at the queue_log. Configure your system to log to mysql and you should be able to get everything you need in realtime. Alex -- _ -- 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On 30/10/2011 05:53 a.m., Raj Mathur (राज माथुर) wrote: On Sunday 30 Oct 2011, Sammy Govind wrote: hmmm so IAX channel is playing with you guys. 1- Cant you guys use SIP, does this happen with SIP trunk as well !? 2- Which version of asterisk are there on both servers. 3- See the output of the command core show file versions in your both asterisk servers. Mainly lookout for IAX channel version. Also try enabling IAX debug and paste the output on console. 1.6.2.9-2+squeeze3 on the SIP server, 1.6.2.9-2+squeeze1 on the Dial server. I doubt if we'll be able to change the architecture of an infrastructure handling up to 450 simultaneous calls for the past 6 months at this stage, so SIP is out. IAX2 has been working beautifully for our needs up to this point, and we need to find a solution that we can integrate into this architecture itself! Incidentally, if anyone's interested, the installation itself is detailed at: http://www.mail-archive.com/ilugd@lists.linux-delhi.org/msg28166.html Regards, -- Raj Sorry if i missed it, but is IAX2 trunked? IF so, perhaps you are running out of bandwidth in your IAX2 trunk. The setting 'trunkmaxsize' defaults to 128000 bytes. From the readme file: ...Once this limit is ; reached, calls may be dropped or begin to lose audio. Depending on the codec in use and ; number of channels to be supported this value may need to be raised, but in most cases the ; default value is large enough. -- Alex -- _ -- 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] Zaptel data mode not supported?
Steve Totaro wrote: Sorry for all the replies, I found the Digium PDF on Data mode. http://www.modulo.ro/Modulo/docs/TE405-410P-user-manual.pdf Good luck getting them to support it though ;) I will post my Sangoma results tomorrow. Thanks, Steve Totaro On Sun, Apr 6, 2008 at 10:49 AM, Steve Totaro [EMAIL PROTECTED] wrote: Sorry, I cannot find the link to the actual Digium link but here are examples from the wiki: http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration http://www.voip-info.org/wiki-Asterisk+Data+Configuration Tomorrow, I will see if data T1 on a Sangoma card is much more simple. If I find the Digium PDF I will post it. Thanks, Steve Totaro On Sun, Apr 6, 2008 at 6:12 AM, Steve Totaro [EMAIL PROTECTED] wrote: Check page 38 of 74. A real pain. Hopefully either Tzafrir is correct with a different distro (Debian)vor Sangoma makes it simple. Thanks, Steve Totaro On Sun, Apr 6, 2008 at 5:47 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Sun, Apr 6, 2008 at 12:23 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Apr 05, 2008 at 10:38:52PM -0400, Steve Totaro wrote: You need to have the kernel compiled specially for it to work. Are you sure? What exactly is needed? I think you need to rebuild the kernel on Centos, but on Debian this happens to be supported in the default kernel. Didn't get to test that support yet, though. Tzafrir, I am not sure actually. Many years ago I was tasked with setting up E1s, one for data and one for voice. There was no definitive guide, but putting *many* pieces together around the web, I came across blog (hotwo back then) and many other pieces on how to recompile the kernel with the correct options, they were not on by default. This was Whitebox or CentOS (RedHat in other words). Never tried on Debian. I am going to try with a Sangoma T1 on Monday, the ./Setup install script makes it look like it should be simple. We shall see. Thanks, Steve Totaro ___ -- 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 Steve: It is simple. ./Setup install and had 2 E1 from different providers up and running in CISCO HDLC mode within 15 minutes. Now to brush up on channel bonding and I'm set to load balance. It's a shame that it couldn't be just that easy with the TE100P. Alex ___ -- 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] Zaptel data mode not supported?
Steve Totaro wrote: Sorry for all the replies, I found the Digium PDF on Data mode. http://www.modulo.ro/Modulo/docs/TE405-410P-user-manual.pdf Good luck getting them to support it though ;) I will post my Sangoma results tomorrow. Thanks, Steve Totaro On Sun, Apr 6, 2008 at 10:49 AM, Steve Totaro [EMAIL PROTECTED] wrote: Sorry, I cannot find the link to the actual Digium link but here are examples from the wiki: http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration http://www.voip-info.org/wiki-Asterisk+Data+Configuration Tomorrow, I will see if data T1 on a Sangoma card is much more simple. If I find the Digium PDF I will post it. Thanks, Steve Totaro On Sun, Apr 6, 2008 at 6:12 AM, Steve Totaro [EMAIL PROTECTED] wrote: Check page 38 of 74. A real pain. Hopefully either Tzafrir is correct with a different distro (Debian)vor Sangoma makes it simple. Thanks, Steve Totaro On Sun, Apr 6, 2008 at 5:47 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Sun, Apr 6, 2008 at 12:23 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Apr 05, 2008 at 10:38:52PM -0400, Steve Totaro wrote: You need to have the kernel compiled specially for it to work. Are you sure? What exactly is needed? I think you need to rebuild the kernel on Centos, but on Debian this happens to be supported in the default kernel. Didn't get to test that support yet, though. Tzafrir, I am not sure actually. Many years ago I was tasked with setting up E1s, one for data and one for voice. There was no definitive guide, but putting *many* pieces together around the web, I came across blog (hotwo back then) and many other pieces on how to recompile the kernel with the correct options, they were not on by default. This was Whitebox or CentOS (RedHat in other words). Never tried on Debian. I am going to try with a Sangoma T1 on Monday, the ./Setup install script makes it look like it should be simple. We shall see. Thanks, Steve Totaro ___ -- 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 Thank you for the replies. It was my understanding that rebuilding the kernel was necessary in 2.4 but everything needed was already included in 2.6 series. My bad I guess. I was trying to find a use for old cards I have, but If i'm going to have to use a Sangoma, I'll just use Vyatta which supports them out of the box. Too bad that the agreement between Vyatta and Digium has not resulted in them supporting Digium cards yet. Alex ___ -- 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
[asterisk-users] Zaptel data mode not supported?
Hello: Have a TE110P laying around and decided to see if I could build a router around it. I've tried compiling several versions of zaptel .1.4.x with the same results. I checked the zaptel changelog and can't find anything about it no longer being supported (or that it ever was for that matter). ztcfg: Zaptel Configuration SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Network HDLC (Default) (Slaves: 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31) 31 channels configured. Changing signalling on channel 1 from Unused to Network HDLC ZT_CHANCONFIG failed on channel 1: Function not implemented (38) dmesg: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.9.2 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :02:06.0[A] - GSI 16 (level, low) - IRQ 209 FALC version: TE110P: Setting up global serial parameters for E1 FALC V1.2 TE110P: Successfully initialized serial bus for card Found a Wildcard: Digium Wildcard TE110P T1/E1 Zaptel networking not supported by this build. make data: make[1]: Entering directory `/usr/src/zaptel-1.4.9.2/menuselect' make[2]: Entering directory `/usr/src/zaptel-1.4.9.2/menuselect' make[2]: `menuselect' is up to date. make[2]: Leaving directory `/usr/src/zaptel-1.4.9.2/menuselect' make[1]: Leaving directory `/usr/src/zaptel-1.4.9.2/menuselect' make -C datamods datamods make: *** datamods: No such file or directory. Stop. make: *** [data] Error 2 Adding datamods to SUBDIR_MODULES in top level Makefile make: CC [M] /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.o /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `fr_lmi_send': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:450: error: `LMI_CISCO' undeclared (first use in this function) /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:450: error: (Each undeclared identifier is reported only once /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:450: error: for each function it appears in.) /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `fr_set_link_state': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:575: error: structure has no member named `bandwidth' /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `fr_lmi_recv': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:646: error: `LMI_CISCO' undeclared (first use in this function) /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:825: error: structure has no member named `bandwidth' /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:829: error: structure has no member named `bandwidth' /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:845: error: structure has no member named `bandwidth' /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `fr_rx': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:878: error: `LMI_CISCO' undeclared (first use in this function) /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c: In function `hdlc_fr_ioctl': /usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.c:1209: error: `LMI_CISCO' undeclared (first use in this function) make[4]: *** [/usr/src/zaptel-1.4.9.2/kernel/datamods/hdlc_fr.o] Error 1 make[3]: *** [/usr/src/zaptel-1.4.9.2/kernel/datamods] Error 2 make[2]: *** [_module_/usr/src/zaptel-1.4.9.2/kernel] Error 2 Am I missing something, or does zaptel.conf.sample need some updating? Alex ___ -- 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] Wildcard TE110P in Mexico
We have several in operation but with isdn and not R2. I know Ive seen emails from people that use them with Telmex and have them operating, albeit with some difficulty. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Cisneros Sent: Friday, September 23, 2005 5:57 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Wildcard TE110P in Mexico Hi I have one question, somebody can tell me if the card TE110P work in mexico, and maybe can tell me the config. Thanks ___ --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] How can i call to a cellphone here in Mexico?
Claudio: In order to receive help from this list, you need to include more information. How are you connecting to the carrier? What are you using as terminals? Softphone? Which one? SIP or IAX2? Hardphone? Brand and model. Contents of your extensions.conf, zapata.conf, and zaptel.conf Are you using AMP to setup your system? These are just a few of the things I can think of that would allow someone to help you. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claudio Canseco Sent: Wednesday, September 21, 2005 12:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How can i call to a cellphone here in Mexico? Hi, I've been trying to dial out to a cellphone, but all my calls get redirected to 066 (the emergency number at my city, like 911) does anyone know how to fix this, any ideas,? does anyone from mexico has done this? Any comment will be highly appreciated, Regards, Claudio ___ --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] ChanSpy
We got it to work by setting the SPYGROUP
variable before every dial command for each group to be monitored and before the
call to ChanSpy by the quality agents. The way I understood the example,
both have to belong to the same SPYGROUP. So for 2
different groups, crm and sales, we use:
[crm-agents-site2]
exten =
_[1-6]XXX,1,SetVar(SPYGROUP=CRM)
exten =
_[1-6]XXX,2,Dial(IAX2/g2/${EXTEN},15,tTwW)
[sales-agents-site2]
exten = _[1-6]XXX,1,SetVar(SPYGROUP=SALES)
exten =
_[1-6]XXX,2,Dial(IAX2/g2/${EXTEN},15,tTwW)
[app-chanspy-crm-agents]
exten = _*53,1,SetVar(SPYGROUP=CRM)
exten = _*53,2,ChanSpy(|g(CRM)q)
[app-chanspy-sales-agents]
exten = _*54,1,SetVar(SPYGROUP=SALES)
exten = _*54,2,ChanSpy(|g(SALES)q)
You could setup a single app which will
prompt you for which group to monitor and set the variable accordingly.
You could even password protect each group to selectively allow access. We
found that you can do the same on the inbound leg of a call to a queue by
setting the variable before the queue command and monitor inbound calls as
well.
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jennifer Hales
Sent: Monday,
September 12, 2005 6:17 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ChanSpy
Hi all,
Does any one know how to make the
g option work with Chanspy? I have done this and it does
not work.
[snoop]
include
= restricted
exten
=756,1,Set(${SPYGROUP}=1)
exten
=756,2,ChanSpy(Agent,qg)
exten
=756,3,Hangup
Regards
Jenn Hales
___
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[Asterisk-Users] IAX2 Problems causing server to hang
Setup is as follows: Server 1 Single TE110p configured for E1 40 SIP softphone clients Voicemail to email on 3 of the extensions Server 2 Single TE110p configured for E1 40 IAX2 softphone clients Voicemail to email on 2 of the extensions Server 2 has random service interruptions with a log file full of channel.c: Dropping voice to exceptionally long queue on IAX2 Solution ranges from ssh to server and restarting * to having to log on the console and reboot the server because there is no network access. Everything points to IAX2. Short of changing all clients to SIP, does anyone have any ideas? Alex ___ --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] Wire Tapping on Asterisk
Chanspy works like a charm if all you want to do is listen to the calls. Only problem is that it's a HEAD only feature at the moment. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Thursday, July 14, 2005 8:30 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Wire Tapping on Asterisk Naw, you're wrong. Look at the Monitor command: http://www.voip-info.org/wiki-Asterisk+cmd+monitor -Original Message- From: Christoph [mailto:[EMAIL PROTECTED] Sent: Thursday, July 14, 2005 5:26 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Wire Tapping on Asterisk On Thu, 2005-07-14 at 17:00 +0800, Ian Bert Tusil wrote: I'm new to asterisk. I would like to ask if there's a feature in asterisk wherein you can monitor ongoing calls, some kinda like tapping into active phone calls? It must have this feature but I do not know where to get some reference to set this up or test this. Can anyone share me some sites as reference? As far as I know there is no feature in Asterisk, but I might be wrong. However, you can use ethereal to tap SIP connections. You simply sniff the SIP connection and after it's done you can decode it and ethereal will output a .au file which contains both sides of the conversation. Also I heared that the Windows tool Cain Able is able to play back SIP converstaions in real time, but I haven't tested that myself. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
