[asterisk-users] Agents, Members and queues
Hi,
I'd like to use queues in Asterisk and I have a few basic questions
(I'm a newbie) about Queues:
1.- What are the differences between Agents and Members (if any)?
2.- I want to implemment a small call center and I think the best way it is by
using dynamic members (agents¿?). In this case, I don't need declare the
members in agents.conf, is it? So, I'd simply declare them in sip.conf and I'd
use "add member" asterisk command. Is it Ok or I'm grong?
3.- I'd like to make an application for that members use call center. I've
thought use some language (java, python...) and work directly with AMI (using a
telnet connection) in order to send commands and receive and processing events
to/from asterisk. Is it a correct way?
Thank you very much in advance.
Mario.
--
Antonio Mario Molina Saorín
web: http://antonio-mario.com
twitter: @a_mario_molina
--
_
-- 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] [NEWBIE] Right dect to buy to use with asterisk
Hello, I need to setup this configuration: - asterisk as IVR; - dect phones. So basically I need a standard set of features: - each dect phone has its extension so I can call it directly; - handover of a call with R key; - if a call is not replied by someone ring all phones. I have little budget. I can choose to buy a fritz!box or a gigasect dect/ip base station. Which one should I buy? Thanks, Mario -- _ -- 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] Problem with wct4xxp - cannot make calls
18:00:40 server kernel: wct4xxp :06:08.0: timing source auto Jul 5 18:00:40 server kernel: wct4xxp :06:08.0: Evaluating spans for timing source Jul 5 18:00:40 server kernel: wct4xxp :06:08.0: span 1 is green : syncpos 1 Jul 5 18:00:40 server kernel: wct4xxp :06:08.0: span 2 is green : syncpos 2 Jul 5 18:00:40 server kernel: wct4xxp :06:08.0: span 3 is green : syncpos 3 Jul 5 18:00:40 server kernel: wct4xxp :06:08.0: span 4 is green : syncpos 4 Jul 5 18:00:40 server kernel: wct4xxp :06:08.0: RCLK source set to span 1 Jul 5 18:00:40 server kernel: wct4xxp :06:08.0: Recovered timing mode, RCLK set to span 1 Jul 5 18:00:40 server kernel: 2G: Got interrupt, status = 000b, CIS = 0088 Jul 5 18:00:40 server kernel: Reg 5 is Jul 5 18:00:41 server kernel: VPM400: Support Disabled Jul 5 18:00:41 server kernel: VPM450: Support Disabled Jul 5 18:00:41 server kernel: Completed startup! Jul 5 18:00:41 server kernel: 2G: Got interrupt, status = 000b, CIS = 0088 Jul 5 18:00:41 server kernel: Reg 5 is Jul 5 18:00:41 server kernel: 2G: Got interrupt, status = 000a, CIS = 0080 Jul 5 18:00:41 server kernel: Reg 5 is Jul 5 18:00:41 server kernel: 2G: Got interrupt, status = 000a, CIS = 0080 Jul 5 18:00:41 server kernel: Reg 5 is Jul 5 18:00:41 server kernel: 2G: Got interrupt, status = 000b, CIS = 0082 Jul 5 18:00:41 server kernel: Reg 5 is Jul 5 18:00:41 server kernel: 2G: Got interrupt, status = 000b, CIS = 0082 Jul 5 18:00:41 server kernel: Reg 5 is Jul 5 18:00:42 server kernel: 2G: Got interrupt, status = 000b, CIS = 0084 Jul 5 18:00:42 server kernel: Reg 5 is Jul 5 18:00:42 server kernel: 2G: Got interrupt, status = 000a, CIS = 0080 Jul 5 18:00:42 server kernel: Reg 5 is Jul 5 18:00:42 server kernel: 2G: Got interrupt, status = 000b, CIS = 0088 Jul 5 18:00:42 server kernel: Reg 5 is Jul 5 18:00:42 server kernel: 2G: Got interrupt, status = 000a, CIS = 0080 Jul 5 18:00:42 server kernel: Reg 5 is Jul 5 18:00:42 server kernel: 2G: Got interrupt, status = 000a, CIS = 0080 Jul 5 18:00:42 server kernel: Reg 5 is Here is system.conf: # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 # echocanceller=mg2,1-15,17-31 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 span=2,2,0,ccs,hdb3,crc4 # termtype: te bchan=32-62 # dchan=47 # echocanceller=mg2,32-46,48-62 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 span=3,3,0,ccs,hdb3,crc4 # termtype: te bchan=63-93 # dchan=78 # echocanceller=mg2,63-77,79-93 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 span=4,4,0,ccs,hdb3,crc4 # termtype: te bchan=94-124 # dchan=109 # echocanceller=mg2,94-108,110-124 # Global data loadzone= us defaultzone = us I appreciate your help in finding out what is wrong with my setup. Thanks! Mario attachment: mmoran.vcf-- _ -- 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] problem with T.38 media headers
Hi Guys, Something I have noticed while dealing with T.38 and re-invites in Asterisk 1.4.22. I have a provider who re-invites with the following sdp (message flow PROVIDER_EQPMT - ASTERISK): . v=0. o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER. s=-. c=IN IP4 CONN_IP_PROVIDER. t=0 0. m=audio 0 RTP/AVP 0. m=image 26858 udptl t38. a=T38FaxMaxBuffer:288. a=T38FaxRateManagement:transferredTCF. a=T38FaxUdpEC:t38UDPRedundancy. The answer coming from asterisk in this case is: . v=0. o=root 3484 3485 IN IP4 CONN_IP_ASTERISK. s=session. c=IN IP4 CONN_IP_ASTERISK. t=0 0. m=image 4653 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:9600. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:200. a=T38FaxMaxDatagram:200. a=T38FaxUdpEC:t38UDPRedundancy. I see a problem here since the number of matched media streams from the offer does not match with the number of matched media streams in reply from asterisk (notice the m=audio 0 RTP/AVP 0 not present in the reply). Please let me know if there are workarounds on this issue, or if this could be a bug on asterisk side. Best regards, Mario Staphorst _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/___ -- 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] problem with T.38 media headers
Hi, I think this is not completely right, The scenario is: Carrier == Asterisk 1.4 == T.38 ATA box. What happends is that the header disappears within the Asterisk server and is not reaching the ATA.I think the SDP headers should be passed through in all circumstances, even if Asterisk 1.4 is only doing T.38 passthrough? Regards, Mario Date: Wed, 27 May 2009 09:44:56 -0400 From: abalas...@evaristesys.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problem with T.38 media headers This is not a problem. Asterisk is under no obligation to offer an audio codec in return. mario staphorst wrote: Hi Guys, Something I have noticed while dealing with T.38 and re-invites in Asterisk 1.4.22. I have a provider who re-invites with the following sdp (message flow PROVIDER_EQPMT - ASTERISK): . v=0. o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER. s=-. c=IN IP4 CONN_IP_PROVIDER. t=0 0. m=audio 0 RTP/AVP 0. m=image 26858 udptl t38. a=T38FaxMaxBuffer:288. a=T38FaxRateManagement:transferredTCF. a=T38FaxUdpEC:t38UDPRedundancy. The answer coming from asterisk in this case is: . v=0. o=root 3484 3485 IN IP4 CONN_IP_ASTERISK. s=session. c=IN IP4 CONN_IP_ASTERISK. t=0 0. m=image 4653 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:9600. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:200. a=T38FaxMaxDatagram:200. a=T38FaxUdpEC:t38UDPRedundancy. I see a problem here since the number of matched media streams from the offer does not match with the number of matched media streams in reply from asterisk (notice the m=audio 0 RTP/AVP 0 not present in the reply). Please let me know if there are workarounds on this issue, or if this could be a bug on asterisk side. Best regards, Mario Staphorst Express yourself instantly with MSN Messenger! MSN Messenger http://clk.atdmt.com/AVE/go/onm00200471ave/direct/01/ ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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 _ What can you do with the new Windows Live? Find out http://www.microsoft.com/windows/windowslive/default.aspx___ -- 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] problem with T.38 media headers
Hi Kevin, Thank you for your reply.I understand that Asterisk is not a SIP proxy, but shouldnt this header be passed on in order to provide proper T.38 passthrough support in this case?As far as i can see is this header really needed to make the T.38 connection successfull, when i setup the call directly to the ATA the reinvite is going fine. Do you have any idea how we can fix this issue? Best regards, Mario Date: Wed, 27 May 2009 10:13:27 -0500 From: kpflem...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problem with T.38 media headers mario staphorst wrote: Carrier == Asterisk 1.4 == T.38 ATA box. What happends is that the header disappears within the Asterisk server and is not reaching the ATA. I think the SDP headers should be passed through in all circumstances, even if Asterisk 1.4 is only doing T.38 passthrough? Asterisk is not a proxy; SIP signaling is never 'passed through'; the two legs of a call are completely separate and Asterisk bridges them together when necessary. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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 _ What can you do with the new Windows Live? Find out http://www.microsoft.com/windows/windowslive/default.aspx___ -- 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] IAX2 - Congestion
Hy all! Your Asterisk server is return this log : *CLI -- Executing Dial(Khomp/B0C0, IAX2/*.*.*.*/9834|30|r) in new stack -- Called *.*.*.*/9834 Mar 14 15:35:40 NOTICE[4212]: chan_iax2.c:2836 auto_congest: Auto-congesting call due to slow response -- IAX2/*.*.*.*:4569-1 is circuit-busy -- Hungup 'IAX2/*.*.*.*:4569-1' == Everyone is busy/congested at this time (1:0/1/0) When someone use the IAX2 trunk. Can anyone helpme? __ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Direct UA to UA RTP connection
Greetings, I have tried with all conceivable means to get my asterisk (called a in this discussion) to have two SIP user agents (called ua1 and ua2 in this discussion running SJPHONE actually) to communicate directly with one another using RTP. No matter what I do, the RTP traffic always goes between ua1 and a and a and ua2, never ua1 to ua2 directly. In my configuration a, ua1 and ua2 are all within the same network with no NAT in between. Here are the asterisk configuration settings I have: Global Nat=never (tried no also) Sip peers Nat=never (tried no also) Canreinvite=yes Once I get ua1 and ua2 to talk directly, I have another question. If a, ua1 and ua2 were all behind different NAT firewalls (ie a is in Boston, ua1 in Toronto and ua2 in San Jose), what would it take to get ua1 to RTP traffic directly to ua2. In this last scenario, ua1 and ua2 are Linksys PAP2T devices. Your expert help is greatly appreciated. Mario ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Choppy sound in voicemail using Asterisk 1.2.11 on CENTOS4 guest on vmware server
I have had no success in getting the voicemail working on Asterisk 1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried with or without ztdummy device, renice -20 on asterisk process and even real-time priority on the host Windows XP box for the vmware process. I am running on an AMD Athlon 64 X2 4600+. The behaviour is when the voicemail answer, the voice sound ok but when asterisk starts saying the digits from the extension, the sound starts becoming very choppy. The voice after the digits is still choppy. Does anyone have a suggestion? The codec that asterisk is using with the softphone I am using is the GSM codec. Please advise, Mario ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP CANCEL NOT WORKING
Hi All. I'm using asterisk 1.2.10, 1.2.13 and 1.4.0-beta3 wth softphone eyebeam 15 I do the following: eyebeam call to PSTN phone 911234567 and asterisk can't create a zap channel sends CANCEL to eyebeam. The log of eyebeam shows this: [06-11-08]16:52:07.415 | Info (debug) RESIP:TRANSACTION | Matching rule for CANCEL :[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.2.54:5060;branch=z9hG4bK4d29449f;rport=5060 Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] To: 621sip:[EMAIL PROTECTED];tag=6626f537 From: 916331591sip:[EMAIL PROTECTED] Call-ID: 0719856da42f542bZGE1NzllOTI3ZGU4NjIwNDhiOTVjOGJkZmFmOTgxNDk. CSeq: 101 CANCEL User-Agent: Asterisk PBX Content-Length: 0 The first line is incorrect, must be CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Where is sip between CANCEL and ':'? Thanks! --- Mario Fdez. Alonso ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to Dial a number with Sangoma PRI card?
I have a Sangoma PRI card configured for E1 line (i.e. 30+1 channels per port) and I'm not quite sure on how the Dial command should performed. I'm using the standard Dial command as if it were a Zap channel. For example Dial(Zap/1/1234) ; Dial 1234 using channel 1 on port 1 Dial(Zap/32/1234) ; Dial 1234 using channel 1 on port 2 (i.e. 32nd zapata entry) Most often than not this works, but sometimes the call fails. However, reading the Asterisk docs, it says that to dial using a PRI card I should use, instead, the following command: Dial(Zap/1-1/1234) ; Dial using channel 1 on port 1 Dial(Zap/2-1/1234) ; Dial using channel 1 on port 2 Which one is correct? Should I dial Zap/32 or Zap/2-1 to go through the 1st channel on port 2? I'm scared of changing my whole dial plan and then discover that, occasionally, things do not work as expected. Please, can someone who has used Sangoma PRI card help me? My Zapata.conf is set as if we had 60+ channels (something similar to this): context = my_context group = 1 [snip...] signalling = pri_cpe switchtype = euroisdn channel = 1-15, 17-31 ; Same, up to channel 62 Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Dial a number with Sangoma PRI card?
That's ok if I want to dial through a group. But, for my specific requirements, I need to dial through a specific channel. I even need to use the ChanIsAvail application to discover which channels are available. Thus, without using a group, which is the correct way to dial through a PRI? Lacy Moore - Aspendora wrote: I dial using groups. Dial(Zap/g1/1234) I'm pretty sure this was taken off of the examples on the Sangoma website. On 9/19/06, *Mario* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have a Sangoma PRI card configured for E1 line (i.e. 30+1 channels per port) and I'm not quite sure on how the Dial command should performed. I'm using the standard Dial command as if it were a Zap channel. For example Dial(Zap/1/1234) ; Dial 1234 using channel 1 on port 1 Dial(Zap/32/1234) ; Dial 1234 using channel 1 on port 2 (i.e. 32nd zapata entry) Most often than not this works, but sometimes the call fails. However, reading the Asterisk docs, it says that to dial using a PRI card I should use, instead, the following command: Dial(Zap/1-1/1234) ; Dial using channel 1 on port 1 Dial(Zap/2-1/1234) ; Dial using channel 1 on port 2 Which one is correct? Should I dial Zap/32 or Zap/2-1 to go through the 1st channel on port 2? I'm scared of changing my whole dial plan and then discover that, occasionally, things do not work as expected. Please, can someone who has used Sangoma PRI card help me? My Zapata.conf is set as if we had 60+ channels (something similar to this): context = my_context group = 1 [snip...] signalling = pri_cpe switchtype = euroisdn channel = 1-15, 17-31 ; Same, up to channel 62 Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Aspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Dial a number with Sangoma PRI card?
ysuf, that's exactly what I'm doing (in Python instead of PHP, but that doesn't matter). However, my question is: should I ask if ZAP/1 is available or if ZAP/1-1 is available? For example: ChanIsAvail(Zap/1Zap/2Zap/3) or ChanIsAvail(Zap/1-1Zap/1-2Zap/1-3) And, once discovered which channel is available, which form of Dial should I use? Should I say: Dial(Zap/2/1234) or Dial(Zap/1-2/1234) yusuf wrote: hi, I did it like this: I wrote a PHP AGI script, that I call from the dial plan. In the AGI I check fwrite(STDOUT,CHANNEL STATUS $currchan \n); fflush(STDOUT); where currchan is ZAP/1 for instance. It returns whether the channel is used. I then pass this back as a variable back to dial plan, and I use that variable to dial. HTH Mario wrote: That's ok if I want to dial through a group. But, for my specific requirements, I need to dial through a specific channel. I even need to use the ChanIsAvail application to discover which channels are available. Thus, without using a group, which is the correct way to dial through a PRI? Lacy Moore - Aspendora wrote: I dial using groups. Dial(Zap/g1/1234) I'm pretty sure this was taken off of the examples on the Sangoma website. On 9/19/06, *Mario* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have a Sangoma PRI card configured for E1 line (i.e. 30+1 channels per port) and I'm not quite sure on how the Dial command should performed. I'm using the standard Dial command as if it were a Zap channel. For example Dial(Zap/1/1234) ; Dial 1234 using channel 1 on port 1 Dial(Zap/32/1234) ; Dial 1234 using channel 1 on port 2 (i.e. 32nd zapata entry) Most often than not this works, but sometimes the call fails. However, reading the Asterisk docs, it says that to dial using a PRI card I should use, instead, the following command: Dial(Zap/1-1/1234) ; Dial using channel 1 on port 1 Dial(Zap/2-1/1234) ; Dial using channel 1 on port 2 Which one is correct? Should I dial Zap/32 or Zap/2-1 to go through the 1st channel on port 2? I'm scared of changing my whole dial plan and then discover that, occasionally, things do not work as expected. Please, can someone who has used Sangoma PRI card help me? My Zapata.conf is set as if we had 60+ channels (something similar to this): context = my_context group = 1 [snip...] signalling = pri_cpe switchtype = euroisdn channel = 1-15, 17-31 ; Same, up to channel 62 Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Aspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP phone with 2 ethernet jacks
We have used both IP501, IP601 (Polycom), Snom 320 and Snom 360. All of them are good phones with very good quality of voice and full of features. However, SNOM phones have a feature (missing from Polycom) that most of our customers really require: with SNOM phones you have leds for presence support that allow you to see which other extensions are busy (through the Asterisk Hint command). If this is important for you, you should really stay with Snom. Guido Hecken wrote: We like the SNOM 360 Phones. They have really good features. Guido -Ursprüngliche Nachricht- Von: Mindaugas Kuprys [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 25. August 2006 09:40 An: asterisk-users Betreff: [asterisk-users] IP phone with 2 ethernet jacks Hi, Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wanted Sipura but they don't have such product. Thanks, Mindaugas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to find which queue member answered a call?
How can I find out, which queue member did answer a given call? I wish, from within my dialplan (extensions.conf) to write a record tying a given incoming call to the (possible) answering queue member. However, I can't find any easy way to get that info, if not using/intercepting AMI (not that easy through extensions.conf) or perhaps extending the Queue app. Am I missing some important point about the Queue app? I'm really surprised that there isn't a more natural way of such a (apparently) simple task as retrieving the answering member of a Queue. Thanks to anyone who might help. Mario. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to find which queue member answered a call?
BJ Weschke wrote: On 8/20/06, Mario [EMAIL PROTECTED] wrote: How can I find out, which queue member did answer a given call? I wish, from within my dialplan (extensions.conf) to write a record tying a given incoming call to the (possible) answering queue member. However, I can't find any easy way to get that info, if not using/intercepting AMI (not that easy through extensions.conf) or perhaps extending the Queue app. Am I missing some important point about the Queue app? I'm really surprised that there isn't a more natural way of such a (apparently) simple task as retrieving the answering member of a Queue. Thanks to anyone who might help. This information is already available via the queue_log Thanks, that might help but it is not what I'm looking for. It won't solve my problem. What I want, from withing my AGI, is to take a specific action depending on who did answer the call. It wouldn't be much feasible to backward read the queue_log file to discover who answered the call, right? It probably would be a quite CPU intensive task. Any other idea on how to better solve my problem? Any undocumented Asterisk Variable for that? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to find which queue member answered a call?
BJ Weschke wrote: On 8/20/06, Mario [EMAIL PROTECTED] wrote: BJ Weschke wrote: On 8/20/06, Mario [EMAIL PROTECTED] wrote: How can I find out, which queue member did answer a given call? I wish, from within my dialplan (extensions.conf) to write a record tying a given incoming call to the (possible) answering queue member. However, I can't find any easy way to get that info, if not using/intercepting AMI (not that easy through extensions.conf) or perhaps extending the Queue app. Am I missing some important point about the Queue app? I'm really surprised that there isn't a more natural way of such a (apparently) simple task as retrieving the answering member of a Queue. Thanks to anyone who might help. This information is already available via the queue_log Thanks, that might help but it is not what I'm looking for. It won't solve my problem. What I want, from withing my AGI, is to take a specific action depending on who did answer the call. It wouldn't be much feasible to backward read the queue_log file to discover who answered the call, right? It probably would be a quite CPU intensive task. Any other idea on how to better solve my problem? Any undocumented Asterisk Variable for that? Yes. In /trunk I've added a MEMBER_INTERFACE variable that's available so you can see who received the call. Thanks, I just checked the trunk version and, in fact, I found the MEMBERINTERFACE (with no underscore) variable. However, the app_queue.c module (from version 1.2.9.1 which I'm using) is quite different from the /trunk. Thus, at the moment, I'll try to get that info (although in an inefficient way) reading backward the queue_log. Then, when I'll migrate to the (stable) Asterisk version with the MEMBERINTERFACE variable available, I'll use tat variable it instead. Do you have any idea in which version of Asterisk will be available the current /trunk version? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting an cellphone to asterisk
Alvaro, you probably need a GSM Gateway. We've been using the one from Topex (http://www.topex.ro) and it works quite well (at least here in Italy). They have models with a single GSM card on board and with two cards. The gateway gets tied either to a Zap (single SIM) or ISDN (double SIM) channel and then you tell to your Asterisk to route the calls to mobile phone through either the ZAP or ISDN channel. Hote this helps. Alvaro Cornejo wrote: Hi Is there a way to connect an Cellphone to asterisk in order to route calls though it?. This is what I want to do: Here is much cheaper to call from cell to cell than from fixed line to cell. So I want to connect a cell to the asterisk box and create a rule to route calls to a cell through the cell connected to the asterisk box. Is it possible? Can I do it with the standard data USB cell-pc or I need a special cable/connection? Did someone worked this? Wich cell brand/model can I use for that? Any tips would be appreciate. Regards Alvaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Festival through AGI can't handle strings longer than 15 chars
Moises, sorry to bother you again. I did some wrong posting to the newsgroup and that's the reason of my (apparent) delay in answering to your latest question. You can now find the answer in the newsgroup. Please, any idea as to the possible cause? If this is a bug (as it seems to me) I'd be glad to try to fix it... but, can you give me a hint to which possible C modules I should look? Just to understand if this is a problem strictly related to Festival or both to Festival and AGI. For your convenience, I report here the important part of my previous post: My CONCLUSION: the error doesn't probably depend on Festival or Exec command. To make the error occur, I simply do this: a) I restart server b) I run the AGI script with whatever text (as wide as I wish) and it will work c) I shorten the text... it still will work d) I then widen again the text: now it won't work! It will work only as long as I shorten the text (or leave it the same length), but not if I widen it. Thanks once more for your support. Mario. Moises Silva wrote: One step more, enable the following in logger.conf console = notice,warning,error,debug,verbose Application app_festival has some interesting debug messages like: ast_log(LOG_DEBUG, Text passed to festival server : %s\n,(char *)data); and that shows in the console the exact test is passed to the festival server. I keep looking into the code trying to find the reason of the behaviour you describe but I havent succed so far. Please report any feedback. Regards On 8/17/06, Mario [EMAIL PROTECTED] wrote: Thanks for your help, Moises. I did activate the AGI DEBUG as you suggested (thanks for that!). However, I'm now only a little bit more sure that I'm passing the right stuff to the Festival command. Following you'll see what I'm passing for the short text (shorter than 15 chars) and for the wider text. As you can see, both the calls seem to work, but for the 2nd I do not hear any sound. At this point, any idea is really welcome. Thanks for your help. *** Short text *** AGI Rx ANSWER AGI Tx 200 result=0 AGI Rx EXEC FESTIVAL Telefono spento -- AGI Script Executing Application: (FESTIVAL) Options: (Telefono spento) == Parsing '/etc/asterisk/festival.conf': Found AGI Tx 200 result=0 -- AGI Script test_command.py completed, returning 0 == Auto fallthrough, channel 'SIP/1-9803' status is 'UNKNOWN' *** Longer text *** AGI Rx ANSWER AGI Tx 200 result=0 AGI Rx EXEC FESTIVAL Telefono utente spento -- AGI Script Executing Application: (FESTIVAL) Options: (Telefono utente spento) == Parsing '/etc/asterisk/festival.conf': Found AGI Tx 200 result=0 -- AGI Script test_command.py completed, returning 0 == Auto fallthrough, channel 'SIP/1-67c2' status is 'UNKNOWN' Moises Silva wrote: Hi Mario. Have you tried to enable AGI debug? CLI agi debug That will show what Asterisk is receiving from your script. Also enable all the debug messages in the logger.conf file for the console Go and try that and post what you see here, and we may be able to help you On 8/17/06, Mario [EMAIL PROTECTED] wrote: I'm having a tough problem when using Festival with Asterisk through AGI: it seems that when I pass more than 15 chars to the Festival command, when from inside an AGI, no sounds (speech) at all is generated. The following (from inside the dialplan) correctly works: exten = 333,1,Answer() exten = 333,2,FESTIVAL(Telefono spento uno) exten = 333,3,Hangup But, when moved from within an AGI, the same Festival command doesn't work: EXEC FESTIVAL Telefono spento uno the symptom is that no text is played, although the return code from command is zero. One important note: if I shorten the text to Telefono spento (i.e. at most 15-chars wide) everything works as expected. I really can't figure out the reason of this weird behavior. What I can do is to exclude some possible reasons: 1. It is not a festival-related problem since when called from the Dialplan everything works as expected. 2. It is not a language-related issue, since I tried this both with English and Italian 3. It is not a missing call to flush()... yes, I added a flush() at the end of my Python-based AGI call 4. It is not a problem related to Python, since I use Python extensively with AGI Does anyone have a hint on what I can do to investigate or solve this problem? Does enyone know if this is a known bug? Thanks in advance, Mario ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Festival through AGI can't handle strings longer than 15 chars
Thanks, Moises. I'll take a look at the code following your suggestion. Then, if successful, I surely post back the (possible) solution on this thread. Moises Silva wrote: Hi Mario, Im sorry to answer so late, but i have been busy. In fact I give a read in the code trying to find the error. You can start looking at source in res/res_agi.c function handle_exec() this function receives the AGI request of EXEC and then call internal function pbx_exec() found in pbx.c, after that, control passes to apps/app_festival.c function festival_exec(), so the final trace would look something like this: handle_exec() in res/res_agi.c pbx_exec() in pbx.c festival_exec() in apps/app_festival.c Good Look, if you found the problem please report back the results :) Regards On 8/19/06, Mario [EMAIL PROTECTED] wrote: Moises, sorry to bother you again. I did some wrong posting to the newsgroup and that's the reason of my (apparent) delay in answering to your latest question. You can now find the answer in the newsgroup. Please, any idea as to the possible cause? If this is a bug (as it seems to me) I'd be glad to try to fix it... but, can you give me a hint to which possible C modules I should look? Just to understand if this is a problem strictly related to Festival or both to Festival and AGI. For your convenience, I report here the important part of my previous post: My CONCLUSION: the error doesn't probably depend on Festival or Exec command. To make the error occur, I simply do this: a) I restart server b) I run the AGI script with whatever text (as wide as I wish) and it will work c) I shorten the text... it still will work d) I then widen again the text: now it won't work! It will work only as long as I shorten the text (or leave it the same length), but not if I widen it. Thanks once more for your support. Mario. Moises Silva wrote: One step more, enable the following in logger.conf console = notice,warning,error,debug,verbose Application app_festival has some interesting debug messages like: ast_log(LOG_DEBUG, Text passed to festival server : %s\n,(char *)data); and that shows in the console the exact test is passed to the festival server. I keep looking into the code trying to find the reason of the behaviour you describe but I havent succed so far. Please report any feedback. Regards On 8/17/06, Mario [EMAIL PROTECTED] wrote: Thanks for your help, Moises. I did activate the AGI DEBUG as you suggested (thanks for that!). However, I'm now only a little bit more sure that I'm passing the right stuff to the Festival command. Following you'll see what I'm passing for the short text (shorter than 15 chars) and for the wider text. As you can see, both the calls seem to work, but for the 2nd I do not hear any sound. At this point, any idea is really welcome. Thanks for your help. *** Short text *** AGI Rx ANSWER AGI Tx 200 result=0 AGI Rx EXEC FESTIVAL Telefono spento -- AGI Script Executing Application: (FESTIVAL) Options: (Telefono spento) == Parsing '/etc/asterisk/festival.conf': Found AGI Tx 200 result=0 -- AGI Script test_command.py completed, returning 0 == Auto fallthrough, channel 'SIP/1-9803' status is 'UNKNOWN' *** Longer text *** AGI Rx ANSWER AGI Tx 200 result=0 AGI Rx EXEC FESTIVAL Telefono utente spento -- AGI Script Executing Application: (FESTIVAL) Options: (Telefono utente spento) == Parsing '/etc/asterisk/festival.conf': Found AGI Tx 200 result=0 -- AGI Script test_command.py completed, returning 0 == Auto fallthrough, channel 'SIP/1-67c2' status is 'UNKNOWN' Moises Silva wrote: Hi Mario. Have you tried to enable AGI debug? CLI agi debug That will show what Asterisk is receiving from your script. Also enable all the debug messages in the logger.conf file for the console Go and try that and post what you see here, and we may be able to help you On 8/17/06, Mario [EMAIL PROTECTED] wrote: I'm having a tough problem when using Festival with Asterisk through AGI: it seems that when I pass more than 15 chars to the Festival command, when from inside an AGI, no sounds (speech) at all is generated. The following (from inside the dialplan) correctly works: exten = 333,1,Answer() exten = 333,2,FESTIVAL(Telefono spento uno) exten = 333,3,Hangup But, when moved from within an AGI, the same Festival command doesn't work: EXEC FESTIVAL Telefono spento uno the symptom is that no text is played, although the return code from command is zero. One important note: if I shorten the text to Telefono spento (i.e. at most 15-chars wide) everything works as expected. I really can't figure out the reason of this weird behavior. What I can do is to exclude some possible reasons: 1. It is not a festival-related problem since when called from the Dialplan everything works as expected. 2. It is not a language
[asterisk-users] Festival through AGI can't handle strings longer than 15 chars
I'm having a tough problem when using Festival with Asterisk through AGI: it seems that when I pass more than 15 chars to the Festival command, when from inside an AGI, no sounds (speech) at all is generated. The following (from inside the dialplan) correctly works: exten = 333,1,Answer() exten = 333,2,FESTIVAL(Telefono spento uno) exten = 333,3,Hangup But, when moved from within an AGI, the same Festival command doesn't work: EXEC FESTIVAL Telefono spento uno the symptom is that no text is played, although the return code from command is zero. One important note: if I shorten the text to Telefono spento (i.e. at most 15-chars wide) everything works as expected. I really can't figure out the reason of this weird behavior. What I can do is to exclude some possible reasons: 1. It is not a festival-related problem since when called from the Dialplan everything works as expected. 2. It is not a language-related issue, since I tried this both with English and Italian 3. It is not a missing call to flush()... yes, I added a flush() at the end of my Python-based AGI call 4. It is not a problem related to Python, since I use Python extensively with AGI Does anyone have a hint on what I can do to investigate or solve this problem? Does enyone know if this is a known bug? Thanks in advance, Mario ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Festival through AGI can't handle strings longer than 15 chars
Moises, follow on and you'll find the exact output that I got from Asterisk once I raised the more detailed debug level as you suggested. I'm sorry, that's quite a long text, but at least I'm sure you have all the info available. However, I now have SOME IMPORTANT notes to add: 1. I'm running Asterisk 1.2.9.1 2. Once I restarted the server, everything worked fine. Thus, it seems that the problem could be solved by just restarting the server. However... read on. 3. Once restarted the server I changed the text inside my Python AGI and the error appeared again. My CONCLUSION: the error doesn't probably depend on Festival or Exec command. To make the error occur, I simply do this: a) I restart server b) I run the AGI script with whatever text (as wide as I wish) and it will work c) I shorten the text... it still will work d) I then widen again the text: now it won't work! It will work only as long as I shorten the text (or leave it the same length), but not if I widen it. I suspect that there is some malloc()ed area (I can't imagine in which C module) that gets successfully narrowed based on the AGI passed text, but never gets enlarged unless the server restarts... Does it seems reasonable? Hope it helps. Note: this is my 2nd reply. Since I didn't see my 1st reply in the newsgroup, I'm now omitting the console log since it is probably useless once I understood the cause of the problem (what I'm missing is how to fix it). I suspect that because there was too much text, my whole reply has been discarded. Moises Silva wrote: One step more, enable the following in logger.conf console = notice,warning,error,debug,verbose Application app_festival has some interesting debug messages like: ast_log(LOG_DEBUG, Text passed to festival server : %s\n,(char *)data); and that shows in the console the exact test is passed to the festival server. I keep looking into the code trying to find the reason of the behaviour you describe but I havent succed so far. Please report any feedback. Regards On 8/17/06, Mario [EMAIL PROTECTED] wrote: Thanks for your help, Moises. I did activate the AGI DEBUG as you suggested (thanks for that!). However, I'm now only a little bit more sure that I'm passing the right stuff to the Festival command. Following you'll see what I'm passing for the short text (shorter than 15 chars) and for the wider text. As you can see, both the calls seem to work, but for the 2nd I do not hear any sound. At this point, any idea is really welcome. Thanks for your help. *** Short text *** AGI Rx ANSWER AGI Tx 200 result=0 AGI Rx EXEC FESTIVAL Telefono spento -- AGI Script Executing Application: (FESTIVAL) Options: (Telefono spento) == Parsing '/etc/asterisk/festival.conf': Found AGI Tx 200 result=0 -- AGI Script test_command.py completed, returning 0 == Auto fallthrough, channel 'SIP/1-9803' status is 'UNKNOWN' *** Longer text *** AGI Rx ANSWER AGI Tx 200 result=0 AGI Rx EXEC FESTIVAL Telefono utente spento -- AGI Script Executing Application: (FESTIVAL) Options: (Telefono utente spento) == Parsing '/etc/asterisk/festival.conf': Found AGI Tx 200 result=0 -- AGI Script test_command.py completed, returning 0 == Auto fallthrough, channel 'SIP/1-67c2' status is 'UNKNOWN' Moises Silva wrote: Hi Mario. Have you tried to enable AGI debug? CLI agi debug That will show what Asterisk is receiving from your script. Also enable all the debug messages in the logger.conf file for the console Go and try that and post what you see here, and we may be able to help you On 8/17/06, Mario [EMAIL PROTECTED] wrote: I'm having a tough problem when using Festival with Asterisk through AGI: it seems that when I pass more than 15 chars to the Festival command, when from inside an AGI, no sounds (speech) at all is generated. The following (from inside the dialplan) correctly works: exten = 333,1,Answer() exten = 333,2,FESTIVAL(Telefono spento uno) exten = 333,3,Hangup But, when moved from within an AGI, the same Festival command doesn't work: EXEC FESTIVAL Telefono spento uno the symptom is that no text is played, although the return code from command is zero. One important note: if I shorten the text to Telefono spento (i.e. at most 15-chars wide) everything works as expected. I really can't figure out the reason of this weird behavior. What I can do is to exclude some possible reasons: 1. It is not a festival-related problem since when called from the Dialplan everything works as expected. 2. It is not a language-related issue, since I tried this both with English and Italian 3. It is not a missing call to flush()... yes, I added a flush() at the end of my Python-based AGI call 4. It is not a problem related to Python, since I use Python extensively with AGI Does anyone have a hint on what I can do to investigate or solve this problem? Does enyone know if this is a known bug? Thanks
RE: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P
I have many problems with this server and two cards TE4110P, because after several minutes one of two cards stays out without sending anyone alarm and then offer a NMI alarm i suppose that it is to cause the sharing IRQ, it´s a ticket for a DIGIUM -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de stoffell Enviado el: Sábado, 20 de Mayo de 2006 07:29 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P On 5/18/06, Remco Barende [EMAIL PROTECTED] wrote: Also the 2850 is *always* sharing IRQ's on every PCI slot, you need to buy a dual port ethernet adapter which will use only one irq to free up an IRQ on another slot. This just totally sucks and irq sharing in a box with only 3 pci slots is totally unnecessary Because we only needed 1 NIC, we disabled the 2nd onboard NIC. That made 1 pci slot free of IRQ sharing, making the system stable and performing very well. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RV: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P
-Mensaje original- De: Mario Montiel [mailto:[EMAIL PROTECTED] Enviado el: Viernes, 26 de Mayo de 2006 10:29 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P I have many problems with this server and two cards TE4110P, because after several minutes one of two cards stays out without sending anyone alarm and then offer a NMI alarm i suppose that it is to cause the sharing IRQ, it´s a ticket for a DIGIUM -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de stoffell Enviado el: Sábado, 20 de Mayo de 2006 07:29 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] DELL PowerEdge 2850 and TE4110P and TE110P On 5/18/06, Remco Barende [EMAIL PROTECTED] wrote: Also the 2850 is *always* sharing IRQ's on every PCI slot, you need to buy a dual port ethernet adapter which will use only one irq to free up an IRQ on another slot. This just totally sucks and irq sharing in a box with only 3 pci slots is totally unnecessary Because we only needed 1 NIC, we disabled the 2nd onboard NIC. That made 1 pci slot free of IRQ sharing, making the system stable and performing very well. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk in FreeBSD
Hello everybody I have a FreeBSD 6.1 box and i would like if exists know issues in asterisk to run in this unix operative sytem I want to know it :) Best regards and thanks in advance Mario ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternative source for Asterisk-IM
Thank's Takayuki Uehara for your information about asterisk-im Takayuki Uehara [EMAIL PROTECTED] Enviado Por: [EMAIL PROTECTED] 16/12/05 01:51 Favor responder a Asterisk Users Mailing List - Non-Commercial Discussion Para:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com cc: Assunto:[Asterisk-Users] Alternative source for Asterisk-IM - I tried to download the Aserisk-IM software from the URL below but the server returns 404 not found response. http://www.jivesoftware.org/wildfire/plugins/asterisk-im.jar Does anybody know any alternative source for downloading Asterisk-IM? Thanks in advance, Ooey -- Takayuki Ooey Uehara [EMAIL PROTECTED] 090-1426-4482, Skype ID: tuehara ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No outgoing sound...sometimes
Verify communication between protocols. SIP ou IAX2. Jason Frisch [EMAIL PROTECTED] Enviado Por: [EMAIL PROTECTED] 13/12/05 00:13 Favor responder a Asterisk Users Mailing List - Non-Commercial Discussion Para:asterisk-users@lists.digium.com cc: Assunto:[Asterisk-Users] No outgoing sound...sometimes - Hi All, I have been having trouble with my asterisk box since last week. It was going fine until then and I can't remember changing anything.. nothing that I haven't put back anyway. The issue is with that about half of the calls received or placed, the outside party cannot hear my voice; I can hear the other end fine. I have checked the logs and nothing is different for the calls that fail. I thought it was the phones, but the messages played from asterisk itself also have the same problem. The native bridge in the below sections seems strange as I though this was disabled with canreinvite=no. denwa*CLI -- Executing Goto(SIP/10.129.46.102-0853ec38, sip|1000|1) in new stack -- Goto (sip,1000,1) -- Executing SetVar(SIP/10.129.46.102-0853ec38, CALLFILENAME=000-20051213-110514) in new sta ck -- Executing GotoIfTime(SIP/10.129.46.102-0853ec38, 18:00-10:00|mon-fri|*|*?24hour|s|1) in n ew stack -- Executing GotoIfTime(SIP/10.129.46.102-0853ec38, *|sat-sun|*|*?24hour|s|1) in new stack -- Executing Dial(SIP/10.129.46.102-0853ec38, SIP/2201SIP/2202|180|tTH) in new stack -- Called 2201 -- Called 2202 -- SIP/2201-afc3 is ringing -- SIP/2202-4367 is ringing -- SIP/2201-afc3 answered SIP/10.129.46.102-0853ec38 -- Attempting native bridge of SIP/10.129.46.102-0853ec38 and SIP/2201-afc3 == Spawn extension (sip, 1000, 4) exited non-zero on 'SIP/10.129.46.102-0853ec38' - conf file: sip.conf [general] port=5060 realm=ocn.ne.jp context=sip [EMAIL PROTECTED]:secret:[EMAIL PROTECTED]/number disallow=all allow=ulaw [number] type=friend host=voip-ca35323.ocn.ne.jp username=username secret=secret fromuser=number fromdomain=ocn.ne.jp port=5060 dtmfmode=inband disallow=all allow=ulaw nat=yes canreinvite=no context=sip [snip] If anybody has any idea where I should look, it would be most appreciated. Jason ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan-capi sound choppy
Hi all. I'm using Debian Sarge with Asterisk 1.0.7.dfsg.1-2 and Asterisk-chan-capi 0.3.5-11 on a P-III 800 with 196MB RAM. The isdn card is AVM B1 isa and the softphone is eyeBeam 1.1 3004t stamp 16741. The audio codec G711aLaw works so fine for me. Other codecs sounds too bad. The problem comes when I use the two B channels of isdn card. The sound is choppy, but if I use only one channel the audio is good. The card is the only card using IRQ 5. The machine at the moment of sound choppy is 70% idle and 55MB RAM free. I had download the source package of Asterisk-chan-capi, and changing AST_CAPI_MAX_B3_BLOCK_SIZE from 160 to 400 the problem of sound choppy is nearly solved. But, that is the way? Thanks. --- Mario Fdez. Alonso Abysal Systems Parque Emp. Las Rozas Jose Echegaray, 5 28230 Madrid Tfl: 916404437 Fax: 916403119 [EMAIL PROTECTED] www.abysal.com --- ___ --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] Programs to parse queue_log
What have other admins done to retrieve detailed call information about
the queue system? Anyone develop their own that they don't mind
sharing?
You can try this perl script it was useful for me. After parsing I do
reports based on generated queue_statistic.csv in Excel...
cut here
#!/usr/bin/perl
#
# Asterisk Queue Analizer
# Uses queue_log to analize call center activities (agents/queues)
# (C) 2005 Mario Spoljar - [EMAIL PROTECTED]
#
# This program is free software, distributed under the terms of the
# GNU General Public License
#
# scriptname: qanalize.pl
#
#
# TO DO:
# - switches for:
# epoch/ordinary output date format
# silent mode
# file to analize
# - cron job description
# - writing to postgresql database
# - rotate output to more then one output file, depending on date
# - xls report output
#
# Change log:
#-
# 12/02/2005
# * added functionality to write last analized record to temporary file
# now script can be called more then once without duplicate output
#
# 10/02/2005
# * improved abilitiy to determine times in case of multiple queues in
and outs
#
#
#
#
# Output fileds:
#
# $outrecord[0] - CallID
# $outrecord[1] - Date Time - equal to Event UniqueID timestamp
# $outrecord[2] - Before Queue - time before call enter queue
# $outrecord[3] - Waiting in Queue
# $outrecord[4] - Talking in Queue
# $outrecord[5] - Answerd yes/no
# $outrecord[6] - Called queue - first queue if more then one
# $outrecord[7] - Answerd by - agent or station
# $outrecord[8] - Caller info - CLID
# $outrecord[9] - Start queue position
# $outrecord[10] - End queue position
# $outrecord[11] - End cause: Abandon, CompleteByAgent, CompleteByCaller
# $outrecord[12] - Queue name - last queue (if call is routed through more
then one queue)
# $outrecord[13] - Total call time (before queue + waiting in queue +
talking)
# $outrecord[14] - Connect time - time when call is cannected
# $outrecord[15] - EnterQueue count how many times call enter queues
# $outrecord[16] - EnterQueue flow - how call was routedthrough queues ex:
q201=all-agents
# $outrecord[17] - Agent session durration - duration in seconds
# $outrecord[18] - Agent Logout time - when agent logout
use Time::Local;
# Variables#
#
# File handles
#
# Which file will be parsed - In file
$file = queue_log;
# Temp file
$tmp_out = tmp_analize_queue_log.txt;
# Out file
$queue_statistic = queue_statistic.csv;
# Max Timestamp value - file
$lastUniqueID = .maxUniqueID;
# Max call duration in seconds (1800 = 30 min)
# in this period of time all call have to be finihed in order to
# parse log correctly (Total call time then max_call_durration)
$max_call_durration = 1800;
#
# Global temporary vairables
#
$tmpoutrecord = ; # previous outrecord for
same call
#
# Default strigs in analized log
#
$AnswerString_caseAbandoned = NO;
$EndCauseString_caseAbandoned = ABANDONED;
$AnsweredBy_caseAbandoned = *NOBODY*;
$AnswerString_caseConnected = YES;
$AnswerString_caseCompleteAgent = YES;
$EndCauseString_caseCompleteAgent = COMPLETE_BY_AGENT;
$AnswerString_caseCompleteCaller = YES;
$EndCauseString_caseCompleteCaller = COMPLETE_BY_CALLER;
$AnswerString_caseTransfer = YES;
$EndCauseString_caseTransfer = TRANSFERED;
$AnswerString_caseAgentCallBackLogoff = SERVICE;
$EndCauseString_caseAgentCallBackLogoff = AGENT_SESSION_TERMINATED;
$AnswerString_caseExitWithTimeout = NO;
$EndCauseString_caseExitWithTimeout = TIMED_OUT;
$AnswerString_caseExitWithKey = NO;
$EndCauseString_caseExitWithKey = PRESSED KEY;
# Main functions
sub prepare_tmp_file {
#
# reareange queue_log, skip evens with uniqueid = 'NONE'
#
# from min_uniqueid to max_uniqueid
#
# queue_log - fileds before :
#
TimeStamp|UniqueID|QueueName|AgentName|Event|[Parameter1|Parameter2|Parameter3]
#
# queue_log - fileds after rearanging:
#
UniqueID|TimeStamp|QueueName|AgentName|Event|[Parameter1|Parameter2|Parameter3]
#
($min_uniqueid, $max_uniqueid) = ($_[0], $_[1]);
open(FH_in,$file) || die cannot open: $!;
open(FH_out,$tmp_out) || die cannot open: $!;
while (FH_in){
#
# reverse filed 0 and 1 in list because we wolud like to sort on
field UniqueID originaly stored on filed 1
#
@list = split(/\|/,$_);
if ($list[0] != NONE) {
#
# dont print rows with UniqueID = NONE - these event belongs to
# restart functions and are not analised there
#
@uniqueid = split(/\./,$list[1]);
if($uniqueid[0] $min_uniqueid $uniqueid[0] = $max_uniqueid){
$tmp = $list[1];
$list[1]=$list[0];
$list[0]=$tmp;
print FH_out join(|,@list);
}
}
}
close(FH_in) || die cannot close: $!;
close
[Asterisk-Users] Re: Asterisk as Cisco Call-Manager - dial out to PSTN
Hi Maron, Thank you for your answer! I use a simple cisco router 2621XM as call manager with the following configuration: interface Loopback79 description ALT-VoIP-Gateway ip address 10.xxx 255.255.255.255 h323-gateway voip interface h323-gateway voip id Ldnxxx ipaddr 10.xxx 1719 priority 120 h323-gateway voip h323-id [EMAIL PROTECTED] h323-gateway voip tech-prefix 301 h323-gateway voip bind srcaddr 10.xxx The structure is Sip-phone SIP Asterisk as call-manager (extension 399) H.323 cisco gatekeeper (extension ) H.323 cisco call-manager (extension 302) E1 PSTN Iif I dial now with the Sip-phone: 302 [PSTN number (handy number, .)] I should be able to telephone the the PSTN of the call manager with the extension 302. It works within cisco devices perfectly but not with asterisk. Can you tell me your experiences and practices?? Thanks a lot!! Mario Hi Mario.What kind of Cisco gateway are you using, I swapped an Cisco Call Manager 4.0 for Asterisk, and am using 12 gateways worldwide for PSTN access. However using SIP, which the gateways (Call Manager Express on 1760 routers) support very well for trunking.I've found that H323 is even buggy between the CME gateways from Cisco.Regards,Maron KristoferssonMario Spendier wrote: Hi all,Im running Asterisk since two days, and its really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me? Im running against closed doors in this problem!!! If I phone over a Cisco call manager it works, so the failure is Asterisk based.-- Executing NoOp(SIP/12345-454d, call for ) in new stack -- Executing Dial(SIP/12345-454d, OH323/ ) in new stack -- H.323 call to with codec alaw -- Called -- H.323 call 'ip$localhost/27230' cleared, reason 24 (Call ended with Q.931 cause) -- Hungup 'OH323/L27230'Thanks a lot!!!Mario ___ Asterisk-Users mailing list Asterisk-Users at 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
RE: [Asterisk-Users] Asterisk as Cisco Call-Manager - dial out to PSTN
Hi Maron, Thank you for your answer! I use a simple cisco router 2621XM as call gateway with the following configuration: interface Loopback79 description ALT-VoIP-Gateway ip address 10.xxx 255.255.255.255 h323-gateway voip interface h323-gateway voip id Ldnxxx ipaddr 10.xxx 1719 priority 120 h323-gateway voip h323-id [EMAIL PROTECTED] h323-gateway voip tech-prefix 301 h323-gateway voip bind srcaddr 10.xxx The structure is Sip-phone à SIP à Asterisk as call-manager (extension 399) à H.323 à cisco gatekeeper (extension ) à H.323 à cisco gateway (extension 302) à E1 PSTN Iif I dial now with the Sip-phone: 302 [PSTN number (handy number, .)] I should be able to telephone the the PSTN of the gateway with the extension 302. It works within cisco devices perfectly but not with asterisk. Can you tell me your experiences and practices?? Thanks a lot!! Mario From: Mario Spendier [mailto:[EMAIL PROTECTED] Sent: Donnerstag, 24. März 2005 13:30 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk as Cisco Call-Manager - dial out to PSTN Hi all, Im running Asterisk since two days, and its really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me? Im running against closed doors in this problem!!! If I phone over a Cisco call manager it works, so the failure is Asterisk based. -- Executing NoOp(SIP/12345-454d, call for ) in new stack -- Executing Dial(SIP/12345-454d, OH323/ ) in new stack -- H.323 call to with codec alaw -- Called -- H.323 call 'ip$localhost/27230' cleared, reason 24 (Call ended with Q.931 cause) -- Hungup 'OH323/L27230' Thanks a lot!!! Mario ___ 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] ser, asterisk and conferencing
Hi ron, Of course you can make meetme, what you need is a zaptel device or, if you haven't any hardware, the ztdummy device. Install it (google), compile asterisk again, define an extension and it should work, more or less ;-))! Greetings, Mario -Original Message- From: ron [mailto:[EMAIL PROTECTED] Sent: Donnerstag, 31. März 2005 16:07 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ser, asterisk and conferencing Hi List, Can I use asterisk to enable call conferencing? I'm using ser for the UA's to register, can I do something like if they dial a certain digits, it will forward it asterisk and use asterisks meetme feature? can i do meetme using only sip? Sorry for my terms, hope you understand my question. Regards, Ron ___ 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
RE: [Asterisk-Users] Ztdummy is Loaded but Asterisk is not using it
Hi, Try to recompile Asterisk and it will work. Greetings, Mario -Original Message- From: RockWater ! [mailto:[EMAIL PROTECTED] Sent: Freitag, 01. April 2005 08:39 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Ztdummy is Loaded but Asterisk is not using it Hello, I have a problem with * on Fedora Core 3 Kernel 2.6. I set up the ztdummy module by following the instructions here http://www.voip-info.org/wiki-Asterisk+timer+ztdummy. The compile worked ok and I edited the files mention in the wiki. Here is a screen grab of what I see when I run lsmod [EMAIL PROTECTED] /]# lsmod Module Size Used by ztdummy 3924 0 zaptel207364 1 ztdummy crc_ccitt 2113 1 zaptel And this is what I see when I do a reload at the astersisk console Apr 1 16:17:18 WARNING[2400]: chan_iax2.c:7311 build_user: Unable to support trunking on user '2277' without zaptel timing I seems like Asterisk is not aware of the presence of ztdummy. Anyone got any suggestions ? Rockwater ___ 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
[Asterisk-Users] Asterisk as Cisco Call-Manager - dial out to PSTN
Hi all, Im running Asterisk since two days, and its really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me? Im running against closed doors in this problem!!! If I phone over a Cisco call manager it works, so the failure is Asterisk based. -- Executing NoOp(SIP/12345-454d, call for ) in new stack -- Executing Dial(SIP/12345-454d, OH323/ ) in new stack -- H.323 call to with codec alaw -- Called -- H.323 call 'ip$localhost/27230' cleared, reason 24 (Call ended with Q.931 cause) -- Hungup 'OH323/L27230' Thanks a lot!!! Mario ___ 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] IRQ headaches
Gary,
I am using TE110P card with [EMAIL PROTECTED] an I had some trouble to setup
corectly, maybe my experiance helps you
Excuse my ignorance here, but I am desperately trying to isolate the IRQ
for
my TE110P card (shown below as t1xxp) Ive gone into my bios and disabled
all
TE110P card should use wcte11xp drivers I am not sure about t1xxp (maybe
someone know if this is same under /proc/interrupts)
1. Edit /etc/init.d/zaptel and add driver for TE110p, and removed ztdummy
from there
MODULES=torisa tor2 wct4xxp wct1xxp wcte11xp wcfxo wcfxs wcusb
RMODULES=wcusb wcfxs wcfxo wcte11xp wct1xxp wct4xxp tor2 torisa
2. Edit /usr/src/zaptel/wcte11xp.c and add some lines to look like:
static struct pci_device_id t1xxp_pci_tbl[] = {
{ 0xe159, 0x0001, 0x71fe, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x79fe, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x795e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x79de, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x797e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0 }
};
3. cd /usr/src/zaptel; make clean; make install
for this asterisk should be down (I suppouse)
4. /etc/init.d/zaptel restart
I am running on an HP Compaq D530s with Fedora Core 1, here is my
I use CentOS (it should be similar)
Mario
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE110P experiance
Hello to all, I would like to ask some Digium TE110P users if they can share experiance about this card. I put in service card yesterday but I noticed following (strange) behaviar: - if I have to reboot my computer my zaptel driver fail to start and produce this error: ZT_SPANCONFIG failed on span 1: No such device or address (6) - to solve this problem I have to power cycle my computer and in all cases this brings up card! - does anybody have any info about this hardware, example there are two LED - what is the meaning of these LEDs. I bought this card and got anly card without any papers (just bill :-( ) Regards, [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip wifi phone access point
Hi guys, which typo of access point you are preffere? Is there any that support roaming between areas without interruption of existing SIP call? ___ 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] Attended xfer
I am running 1.0.5 and can happily blind xfer from extension to extension, but I can't blind xfer. I have read various snippets about #2 or #8 or other such key combos, but nothing seems to let me do attended xfer. From xlite I can blind xfer without problem but no attended xfer. For attendant transfer you should use CVS Head, in Asterisk stable is not implemented that feature! ___ 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] multiple nics and internet
Hi all, recently i've posted a request about a big problem i have. I was trying to configure asterisk iax to serve a double ADSL connection and internal network, but even if i've used the default configuration for bindaddr ( 0.0.0.0 ) i had a very strange behaviour. here is my conf.: from ifconfig : eth0 : 192.168.3.1 eth1 : 192.168.4.1 defualt gateway 192.168.3.254. from iax.conf : bindaddr = 0.0.0.0 Well, when i use the first address for iax registration everything goes fine, but when i try to use the second ip address, i will receive a reg ack from the first ip address and my client discard it ( right !! ). client server x.x.x.x REG REQ 192.168.3.1 x.x.x.x REG ACK 192.168.3.1 OK ! x.x.x.x REG REQ 192.168.4.1 x.x.x.x REG ACK 192.168.3.1 NOK! I've made a full search for a solution of this problem and i found few informations ( i hope correct ) 1. Asterisk uses kernel 2.0 based routing ( strange ! ) and it reach the destination based on default gateway. So if i use the same subnet of the second ip address it should work, else it uses the first address that is directly connected to the default gateway. 2. Someone point me to the use of iproute2 as a packet shaper. But if i've understood well, it can only use the destination address of the client and not the source address of the server for routing decision. See below : ip rule add to x.x.x.x lookup Table 1 ( where table 1 has a different default gw ) This is possible but useless when i have internet in the other side ( i couldn't predetermine all the class of subnet ! ) This is not possible ip rule add from 192.168.4.1 lookup Table 1 ( because the source address has to be assigned before the routing decision ) 3. Someone else said that the only solution is to install a second asterisk server to serve the second ADSL link and bind themselves with iax trunk. At the end of this long listing you would know if i've had a solution ? My answer is yes ! here is how. look at this simple iptables command iptables -t nat -A POSTROUTING -i eth2 -o eth0 -d 192.168.4.1 -j SNAT --to-source 192.168.4.2-192.168.4.253 eth2 is the ADSL connection eth0 is the Asterisk connection With this command i force the source ip address of a client from internet to be natted to a dynamic ip address in the right subnet and thus asterisk would use the right ip address to send packet back. Simple !! The only thing to remember is to assign the ip addresses of the pool to the mac-address of the firewall with this command arp -f /etc/ethers where /etc/ethers is: x.x.x.2AA:BB:CC:DD:EE:FF pub x.x.x.3AA:BB:CC:DD:EE:FF pub ... x.x.x.253AA:BB:CC:DD:EE:FF pub if a class C is not enough for you ( you're a lucky boy ) simply use a class B or higher. I've made some testing with iax2 ( it hasn't NAT issue ) and it works fine. If someone else has a better solution Id be happy to hear from you, that's my best. Mario Hopefully it will be useful to someone else. ___ 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] fax over tdm400p
My solution was to use app_rxfax() and some glue to have faxes automatically converted to PDFs and placed in a samba share. Could you describe more detaily how this could be done. I plan to do similar thing, so I would like to know which biniries i have to had todo the same, and to have feeling what other configuration on linux box should be done except * config. I'm not very familiar with samba so I will apreciate if you share your experiance just to start from this point. ___ 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] iax.conf bindaddr parameter not working
Hi, I'm trying to configure a dual homed asterisk server with iax accepting connection on all address. from iax.conf.example bindaddr = 0.0.0.0 Address to bind to ( all addresses on machine ) but if i register a client using the second ip address i will receive the response from the first ip address and obviously the client discard this. let me explain more: Client : 192.168.0.4 Server: 192.168.0.1 - 192.168.0.2 Reg Req : src ( 192.168.0.4) -- dst ( 192.168.0.2 ) Reg Ack : src ( 192.168.0.1) -- dst ( 192.168.0.4 ) invalid !! rejected from the client. I've missed something ? Since it's not normal tcp/ip protocol behaviour, can i consider it a bug ? Can i get a workaround ? Thanks in advance Mario ___ 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] Can't initiate a call with X-Lite.
No. You should work on configuring xlite to register with asterisk. In the xlite Sip Proxy menu, you will need a User Name, Password, Sip Proxy, and Domain/Realm defined to match entries in your sip.conf definitions. to which entry have to corespond Domain/Realm parameter in X-lite ___ 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] agent with queues remain unavailable during transferred call
I've had same problem, but I realised that problem was in fact that my agents used * just for handling incoming queue call and my agent phones have been registered on another legacy PBX (Alcatel 4400) interconected with * through ISDN PRI. Because of that transfer function is handled on legacy PBX (Alcatel) and Asterisk does not 'know' if agent talks to callee or if I transfer incoming call. Do you using some other PBX connected to Asterisk PBX? That may be the case. My topology loks like: +--+ +-+ --PSTN-PRI| ALCATEL |PRI-| ASTERISK| ++-+ +-+ ^ | ¡ ++-+ | AGENT | +--+ This kind of topology were used because: * agents was used their station on Alcatel before * through Asterisk I added some additional features to my call center without need to pay expensive licences to Alcatel * I need functionality of billing application connected to Alcatel PBX Mario Spoljar [EMAIL PROTECTED] [EMAIL PROTECTED] wrote on 03/01/2005 15:53:16: Hi, I'm seeing something I'd like suggestions on: I have a queue with agents that log in using agentcallbacklogin. The extension that is logged in with is a Local channel. Now, if a call comes in to the queue and is handled by an agent (in our case using Cisco 7960 SIP phones) and transferred (attended) to another extension, the agent remains unavailable during the remains of the call. Using show agents gives this: 103 (TIC 3) logged in on MGCP/aaln/[EMAIL PROTECTED] talking to Zap/20-1 (musiconhold is 'default') As you can see, the Agent is shown with the transferred call, and is unavailable for new calls. However, the phone _is_ on hook and free. I am using a 1.0.2. version (bri-stuff rc2b) Any suggestions are welcome. Florian ___ 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
Re: [Asterisk-Users] Sphinx
A great universe to explore... I tried some 6 months ago but there isn´t any great voice project in portuguese (Brazilian)... and CMU release a test code to windows... sphinks + festival + portuguese If you have any news about shpinx+asterisk please let us know... Happy new year, Mario - Original Message - From: Barry Porch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 03, 2005 6:05 PM Subject: ADV: [Asterisk-Users] Sphinx Is anyone doing anything useful with Sphinx? There is a small amount of info on the wiki to help with implementing it but I'm curious if anyone is actually using it. Barry ___ 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
RE: [Asterisk-Users] CallerID in Queue
I suppouse that you are using AgentLogin application, try instead to use
AgentCallbackLogin. Here is example how I do call center application on
my site:
[extension.conf]
---
[macro-agent-login]
; Agent login
; ${ARG1} - Caller nubmber - same as agent number
exten = s,1,AgentCallbackLogin(${ARG1}|[EMAIL PROTECTED])
[macro-agent-logoff]
; Agent logoff
; ${ARG1} - Caller nubmber - same as agent number
exten = s,1,AgentCallbackLogin(${ARG1})
[macro-SetCLIDprefix]
; ${ARG1} - A number
; add prefix for recognision of called destination when sharing agents
;
; ex. ** before CLID user called Helpdesk call queue
;
exten = s,1,NoOp(OriginalCLID:${CALLERIDNUM})
exten = s,2,GotoIf($[${CALLERIDNUM:0:1} = *]?s|4:s|3)
exten = s,3,SetCallerID(${ARG1}${CALLERIDNUM})
exten = s,4,NoOp(ChangedCLID:${CALLERIDNUM})
exten = s,5,Goto(${MACRO_CONTEXT},${MACRO_EXTEN},$[${MACRO_PRIORITY} +
1]);
[callcenter]
; define how to call Zap agents through PRI
; there should be listed all agent destitantions
exten = 3991,1,Dial(Zap/g1/3991)
exten = 3992,1,Dial(Zap/g1/3992)
exten = 3993,1,Dial(Zap/g1/3993)
[CC_HelpDesk]
;
; before A number add ** like identifier that call is originated to
call center -
; usefull when you share agents with more call queues, then agent can
see prefix
; so it can recognise to whome is call placed
;
exten = s,1,ResponseTimeout(15)
exten = s,2,Wait(2)
exten = s,3,Answer
exten = s,4,Playback(GBS-CC/10); Play 'You
reached GBS IT call center'
exten = s,5,SetMusicOnHold(default)
exten = s,6,Macro(SetCLIDprefix,**)
exten = s,7,DigitTimeout(5)
exten = s,8,Queue(hd-q|tn|30) ; r- ring
instead of moh,
; t- transfer alowed,
; n- after
timeout will exit this application and go to the next step
exten = s,9,Background(GBS-CC/2) ; Play 'All
operators are busy...press 1 to leave message, 2 to keep waiting'
exten = 1,1,Voicemail(s5666)
exten = 2,1,Goto(s,8)
exten = t,1,Background(Attendant/zauzeti) ; What to
do after Timout set in s,1
exten = i,1,Goto(CC_HelpDesk,s,9) ;
[default]
; 5670 - prefix to agent logon
exten = 5670,1,Macro(agent-login,${CALLERIDNUM:3}) ; you can log in
yust from your own station, exaple My CLID= 3723991, strip first 3
digits,
; pass to macro
just 3991 (agent station number)
; 5671 - prefix to logoff
exten = 5671,1,Macro(agent-logoff,${CALLERIDNUM:3}); call logoff
macro with my number equal to agent number, after prompt input password,
press twice #
; -
; HD call queue
; extension to call my Help desk call queue
; -
exten = 5666,1, Goto(CC_HelpDesk,s,1)
[queues.conf]
-
[hd-q]
; managed calls from group HD
;
; Notice - if using following notationa:
; member = Agent/@1 - roundrobin strategy doesnot work, you should put
each agent in configuration separately
;
!!
;
music = default
strategy = random
timeout = 15
maxlen = 5
; group 1 (hd-main)
member = Agent/3991
member = Agent/3992
member = Agent/3993
member = Agent/3994
member = Agent/3999
; group 4 (hd-backup)
member = Agent/2170
member = Agent/2314
member = Agent/2603
member = Agent/2858
member = Agent/2216
member = Agent/2864
member = Agent/2688
member = Agent/2701
member = Agent/2952
[agents.conf]
;
;
..
group=1
agent = 3991,1234,HD-3991 ; 3991
agent = 3992,1234,HD-3992 ; 3992
agent = 3993,1234,HD-3993 ; 3993
agent = 3994,1234,HD-3994 ; 3994
agent = 3999,1234,HD-3999 ; 3999
..
Hope it will help you...
Mario Spoljar
IT TO Telecommunications
GBS IT
Croatia
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rolland
Wong
Sent: Monday, September 20, 2004 10:37 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] CallerID in Queue
How can I bring the Caller ID when the calls enter call queue and answer
by X- lite or kphone?
I've tried many configuration but no luck that it only shows the
AgentLogin's exten..
Thanks!
R Wong
The information transmitted is intended only for the person or entity to
which it is addressed and may contain confidential and/or privileged
material.
Any review, retransmission, dissemination or other use of, or taking of
any action in reliance upon, this information by persons or entities
other than the intended recipient is prohibited.
If you received this in error, please contact the sender and delete the
material from any computer.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
This e-mail, and any attachments, may contain
[Asterisk-Users] Re: Fedora2 and Kernel 2.6 again!
I'm a newbie, but I found this information that works for Fedora2: You need to make a symlink /usr/src/linux-2.6 ... you can reference it to /lib/modules/2.6.5-1.358/build/ (I think you don't need to compile your kernel for this)... or you can reference it to /usr/src/linux-2.6.5-1.358/ (but you need to compile your kernel) There is additional information at README.linux26 Mario Velasco _ Do You Yahoo!? Información de Estados Unidos y América Latina, en Yahoo! Noticias. Visítanos en http://noticias.espanol.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk] help with extension switching
Return Receipt Your Re: [Asterisk] help with extension switching document : was Mario Maqueda/DEV/SA/DuPont received by: at: 07/24/2003 12:29:27 AM ZW3 This communication is for use by the intended recipient and contains information that may be privileged, confidential or copyrighted under applicable law. If you are not the intended recipient, you are hereby formally notified that any use, copying or distribution of this e-mail, in whole or in part, is strictly prohibited. Please notify the sender by return e-mail and delete this e-mail from your system. Unless explicitly and conspicuously designated as E-Contract Intended, this e-mail does not constitute a contract offer, a contract amendment, or an acceptance of a contract offer. This e-mail does not constitute a consent to the use of sender's contact information for direct marketing purposes or for transfers of data to third parties. Francais Deutsch Italiano Espanol Portugues Japanese Chinese Korean http://www.DuPont.com/corp/email_disclaimer.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
