Re: [Asterisk-Users] OT: MAX TNT and PRI calling name (CNAM) facility message
For the record/archives, I tested with asterisk sending calls to the MAX with the Display IE in the SETUP message and the MAX still doesn't send the name in headers. Another helpful soul pasted me some config. I'll keep at it and report my findings here, at least for the bots to index. My provider cannot provide CNAM lookup results in the SETUP message, only locally configured names in the DMS-100 (Centrex style). On 6/23/05, Matt Fredrickson [EMAIL PROTECTED] wrote: On Thu, Jun 23, 2005 at 12:20:34AM -0600, Kevin Blackham wrote: Does anyone have a MAX/APX with working ingress PRI calling name? I recently acquired a MAX TNT on the cheap and it's integrating fine except for one thing. In the 11.0.0 release notes, it is stated that ISDN calling name will, if present and permitted by presentation flags, be added to the From: and Remote-Party-ID: headers of the INVITE. I'm not able to make this happen. Pcap captures show it is indeed in neither header, and I suspect the MAX is sending the INVITE before it receives this data. Debug traces show it does receive the message, but due to limitations of the CLI, I cannot correlate whether it's received before or after the INVITE is dispatched. It works great direct to Asterisk (of course) via TE410P on the same NI-2 spans. My FACILITY message that contains the CNAM wanders in from 100 to 400ms after the initial SETUP. I can't seem to find any way to get the MAX to stall for a half-second before invoking the INVITE (if that's even the issue). Is my provider too slow? Is there another valid way for CNAM to be provided during the SETUP message, assuming my provider can stall the call setup until the SS7 query is returned? (google for Q.931 docs not helping me much there either) That's one of the (many) ways that caller name is provided. In fact, it's pretty much the most common way that I've seen for ISDN PRI. I don't know if you're provider supports it, but sometimes you can get it in the SETUP message. I'm not sure what level of control they have though. Matthew Fredrickson ___ 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] OT: MAX TNT and PRI calling name (CNAM) facility message
Does anyone have a MAX/APX with working ingress PRI calling name? I recently acquired a MAX TNT on the cheap and it's integrating fine except for one thing. In the 11.0.0 release notes, it is stated that ISDN calling name will, if present and permitted by presentation flags, be added to the From: and Remote-Party-ID: headers of the INVITE. I'm not able to make this happen. Pcap captures show it is indeed in neither header, and I suspect the MAX is sending the INVITE before it receives this data. Debug traces show it does receive the message, but due to limitations of the CLI, I cannot correlate whether it's received before or after the INVITE is dispatched. It works great direct to Asterisk (of course) via TE410P on the same NI-2 spans. My FACILITY message that contains the CNAM wanders in from 100 to 400ms after the initial SETUP. I can't seem to find any way to get the MAX to stall for a half-second before invoking the INVITE (if that's even the issue). Is my provider too slow? Is there another valid way for CNAM to be provided during the SETUP message, assuming my provider can stall the call setup until the SS7 query is returned? (google for Q.931 docs not helping me much there either) I know this isn't the place for Ascend/Lucent MAX discussion, but there doesn't seem to be anything active out there. I'm looking for a mail list/newsgroup/community if there is one still alive. ___ 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] newlines in application data strings (e.g. userevent)
exten = s,9,UserEvent(AgentMoreTime,Agent: ${agent}\r\nUntil: ${wrapupat});
Fragment \r\n parses into rn. \\r\\n turns into \r\n (uninterpreted).
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Re: [Asterisk-Users] Multiline / Console / Receptionist phone
Have you configured the 'break' key in the web ui? Not exactly an intuitive option, but that's the 'xfer' key. On Tue, 14 Dec 2004 09:06:14 -0800, Tracy R Reed [EMAIL PROTECTED] wrote: The only thing that does not seem to work on the snom phone is transfers which is unfortunately a big problem. Attended transfers work (answer incoming call, put on hold, call destination party and advise of the call, hang up, call that was on hold is now transferred) but the blind transfer where you push the transfer button on the phone does not seem to work. I am still trying to figure this one out. ___ 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
Re: [Asterisk-Users] New PRI with DID in US?
On Fri, 10 Dec 2004 17:26:48 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
Just turned up a new PRI with DID's in the US. I'm receiving 5 digits
of the DID numbers as I requested.
Assuming I have 100 DID numbers but only define 50 of those in
extensions.conf, is there an easy way to send the incoming calls
for the 20 undefined numbers to a common resource (ivr, operator,
or canned message) without having to define each one?
I handle my DIDs with a macro. A DBget fetches a target for Goto. If
the key doesn't exist, it jumps to a hangup macro that can either drop
with PRI_CAUSE=1 (invalid) or play Zapateller and ss-noservice.gsm
twice, then hang up with PRI_CAUSE=31, depending on how you want it to
work. Of course, don't answer first, and if you do Playback(),
remember the noanswer option and play a silence/1 first..
Some samples from database show (numbers changed to protect the
guilty, and I receive full 10 digits)
/DID/9001235900 : mainmenu|s
/DID/9001235904 : 104
/DID/9001235917 : 117
/DID/9001235939 : 139
/DID/9001235942 : 142
/DID/9001235970 : 170
/DID/9001235949 : disa|s
[from-pstn]
exten = _NXXNXX,1,Macro(did,${EXTEN});
[macro-did]
exten = s,1,DBget(target=DID/${ARG1})
exten = s,2,Goto(${target},1); db should not include the priority
exten = s,102,SetVar(PRI_CAUSE=1); tells telco to play discon message
exten = s,103,Hangup;
; target can be simply an extension: my stdexten does more DBget to find channel
exten = _1XX,1,Macro(stdexten,${EXTEN})
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[Asterisk-Users] -p real time priority and -U together
When I start asterisk with 'asterisk -pU asterisk -G asterisk', I can get it up and running as non-root with priority. However, if I restart from CLI, it exits complaining about being unable to set prio. Clearly this is because it's non-root. Is this an impossible scenario? ___ 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
Re: [Asterisk-Users] Two zaptel T1 cards: no clock from one
On Sat, 4 Dec 2004 21:18:50 -0600, Rich Adamson [email protected] wrote: Help me understand what you mean by neither is providing clock. By definition, every single T1 provides clocking within the transmit side of a T1. Its embedded in the data stream and you can't turn it off. Are you talking about clock sync? First, In reply to Lyle, framing/line coding match, and the span 5 is in the dmesg, but the wct1xxp doesn't spit out SPAN x and it's mixed in: Found a Wildcard: Digium Wildcard T100P T1/PRI ...Using ESF/B8ZS coding/framing You're absolutely right, so I must be thinking of clock sync, and I'm not clear on recall what I saw from the channel bank. From the T100P though I am sure I saw zero hertz rx. I'll verify again Monday. Anyway, I'll be wrangling Digium again Monday (support was unresponsive Friday). Regardless of other issues, I expect it should be clearing alarms when I stick a loopback plug in. ztcfg (snipped a bit): pbx:~# ztcfg -vv Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 5: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: (snipped out 24,48,72 = D, 1-23,25-47,49-71, 73-94 = B) Channel 95: Individual Clear channel (Default) (Slaves: 95) Channel 96: D-channel (Default) (Slaves: 96) Channel 97: FXO Kewlstart (Default) (Slaves: 97) Channel 98: FXO Kewlstart (Default) (Slaves: 98) Channel 99: FXO Kewlstart (Default) (Slaves: 99) Channel 100: FXO Kewlstart (Default) (Slaves: 100) Channel 101: FXO Kewlstart (Default) (Slaves: 101) Channel 102: FXO Kewlstart (Default) (Slaves: 102) Channel 103: FXO Kewlstart (Default) (Slaves: 103) Channel 104: FXO Kewlstart (Default) (Slaves: 104) Channel 105: FXO Kewlstart (Default) (Slaves: 105) Channel 106: FXO Kewlstart (Default) (Slaves: 106) Channel 107: FXO Kewlstart (Default) (Slaves: 107) Channel 108: FXO Kewlstart (Default) (Slaves: 108) 108 channels configured. zttool (sucky paste): x OK TE410P (PCI) Card 0 Span 1 a x x OK TE410P (PCI) Card 0 Span 2 # x x OK TE410P (PCI) Card 0 Span 3 a x x OK TE410P (PCI) Card 0 Span 4 a x x RED Digium Wildcard T100P T1/PRI Card 0 a x x xCurrent Alarms: Red Alarm x x x xSync Source:Internally clocked x a x x xIRQ Misses: 0 x a x x xBipolar Viol: 0 x a x x xTx/Rx Levels: 0/ 0 x a x x xTotal/Conf/Act: 24/ 12/ 0 x a x x x 112lqqkx a x x x123456789012345678901234x Back xx a x x xTxA mqqjx a x x xTxB x a x x xTxC x # x x xTxD x x x xlqqkx x x xRxA x Loop xx x x xRxB mqqjx x x xRxC x x x xRxD x ___ 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
Re: [Asterisk-Users] Two zaptel T1 cards: no clock from one
Oh, you got to be kidding me. :) I've removed zaptel, wct1xxp, wct4xxp completely, and reloaded in the same order as they are loaded, alternate different ways, etc. No help. After trying only wct1xxp with 12 channels and seeing alarms clear, something stray was left saying span 5 was 'UNCONFIGURED' (with only 1 span in zaptel.conf). So, I rebooted ro see if I could reset some phantom state, and viola! The damn thing is just fine now. Thanks for your help, all. I'll see if I can recreate this again and file a bug report, if so. ___ 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
Re: [Asterisk-Users] Two zaptel T1 cards: no clock from one
Yeah, proper crossover cable. I've eliminated all cabling issues with the T1 analyzer. I get a full and accurate pattern back when I test from the cable end where it would have been connected into the T100P, with the channel bank in loopback. The main symptom is that when I hook the analyzer directly to either the channel bank or the T100P, neither is providing clock. I could have the channel bank supply one, but I will have fax/modem calls bridged between the two PCI cards, so a common clock is best. The most disturbing thing is that the T100P, as the only card in a system, provided clock just fine. There was a thread last month in -dev about being unable to use common clock source across cards. Is this related? How can one cause zaptel to provide ref clock? Should I be seeing 1000 interrupts/sec on any and all TDM cards? On Fri, 03 Dec 2004 23:59:22 -0500, [email protected] wrote: The cable should be cross-connect 1-4, 2-5 each way. Is it? ___ 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] Two zaptel T1 cards: no clock from one
List, I have a TE410P (T1 mode, all PRI) and a T100P (fxoks, for fxs channel bank). I cannot seem to get the T100P to send any clock to the channel bank. I prefer that it use the same clock source as the TE410P, but it doesn't matter if it's not in sync just as long as it's there. The TE410P is configured 3x pri_cpe, 1x pri_net. The three cpe go to XO Sonus switch, the net to legacy PBX. Clock is received from telco, old PBX receives clock from zaptel card, everything's green there, but the other card, the T100P, seems to not send any timing at all, as verified by our T1 analyzer, and is persistently in red alarm. In fact, even if I stick a loopback plug in the T100P, the alarm persists (loopback causes a result in the TE410P). The T100P and channel bank were just pulled from another working * box, and the configuration is nearly identical, except it was the only T1 interface. System: Supermicro dual Xeon 2.4, both cards on same PCI bus. Cards: one T100P, one TE410P. Config: spans 1-4 for quad card (module loaded first), span 5 is single port card Channel bank: Access Bank II, 12 FXS Info dumps (some snipped for brevity) lspci (snipped, these are the only devices on bus 5): :05:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface :05:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev 01) cat /proc/interrupts (odd, shouldn't the T100P be generating 1000 ints/sec?): CPU0 CPU1 CPU2 CPU3 0: 20727 0 18569488 0IO-APIC-edge timer 9: 0 0 0 0 IO-APIC-level acpi 28: 177705 0 0 0 IO-APIC-level eth0 29: 9282 0 0 0 IO-APIC-level eth1 72: 35845 0 0 0 IO-APIC-level dpti0 100:102 0 0 0 IO-APIC-level t1xxp 104: 18342391 0 0 0 IO-APIC-level t4xxp lsmod: Module Size Used by wct1xxp17568 0 wct4xxp70048 0 zaptel226436 222 wct1xxp,wct4xxp e1000 87348 0 crc_ccitt 3072 1 zaptel zaptel.conf: span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs span=3,3,0,esf,b8zs span=4,4,0,esf,b8zs span=5,0,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 bchan=49-71 dchan=72 bchan=73-95 dchan=96 fxoks=97-108 #fxoks=109-120 loadzone = us defaultzone=us asterisk/zapata.conf: [channels] language=en echocancel=yes echocancelwhenbridged=no echotraining=yes echotraining=800 immediate=no ;--pstn-- context=from-pstn signalling=pri_cpe switchtype=dms100 group = 1 channel = 1-23,25-47,49-71 ;--pri to pbx-- signalling=pri_net switchtype=dms100 group = 3 channel = 73-95 ;--channel bank-- context=fax+modem signalling=fxo_ks channel = 97-108 a snippet from dmesg: ACPI: PCI interrupt :05:03.0[A] - GSI 104 (level, low) - IRQ 104 Found TE410P at base address f8401000, remapped to f9b98000 TE410P version c01a009b FALC version: 0005, Board ID: 00 registers snipped TE410P: Launching card: 0 TE410P: Setting up global serial parameters Found a Wildcard: Wildcard TE410P-Xilinx ACPI: PCI interrupt :05:02.0[A] - GSI 100 (level, low) - IRQ 100 Framer: DS21552, Revision: 3 (T1) Found a Wildcard: Digium Wildcard T100P T1/PRI Registered tone zone 0 (United States / North America) TE410P: Span 1 configured for ESF/B8ZS SPAN 1: Primary Sync Source TE410P: Span 2 configured for ESF/B8ZS SPAN 2: Secondary Sync Source TE410P: Span 3 configured for ESF/B8ZS SPAN 3: Tertiary Sync Source TE410P: Span 4 configured for ESF/B8ZS SPAN 4: Quaternary Sync Source Using ESF/B8ZS coding/framing Calling startup (flags is 4099) ___ 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
Re: [Asterisk-Users] Playing reveived message WAV file
Look for the 'play' binary. It has never failed me in playing a wave (gsm, pcm, etc) from asterisk. It's an older OSS (/dev/dsp) binary, so you'll need OSS compatibility. On Thu, 25 Nov 2004 20:12:14 -0700, Joseph [EMAIL PROTECTED] wrote: After somebody records a message asterisk notifies me and encloses the WAV file. Though I'm not sure if this is a WAV format. I can not play it. According to the file specification it is: msg.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz How to play received message? ___ 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
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
I have a 200 and the hint() stuff works fine for indicating status of any channel (including Agent channels). The Snom subscribes to asterisk at whatever url you put in there, then * will send notify events when the dialog state changes. It's not quite a shared-line (at least the way I understand it) but it does dial the extension and show status. Adding the extended keypad to a 220 is just 'more buttons' that are all configurable the same, unlike the Cisco keypad which doesn't do SIP. I'm also working on a receptionist panel that I intend to operate with a touchscreen LCD (probably a 15, there are plenty that have X support). It'll not only subscribe to dialog state of asterisk channels, but will also subscribe to SIP presence (Polycom phones in our case), so she knows if they're on DND, or whatever they set their status to. On the back end, there's an XMLRPC daemon that tracks all this state and abstracts the dirty work of transfers, etc, via the manager port. It also will be doing a number of other things for our in-house Java call center apps. I chose to go this route instead of just the Snom since she would rather have it on a PC anyway (much easier to find your target, can change to suit her), and mainly since DND on SIP phones simply bounces the call as busy without * being able to tell the receptionist not to bother in the first place. Also, transfers will work better this way since it's going to be a native * method, and not SIP refer, with options for vmail busy, vmail unavail, blind, attended or camp-on (my dial plan is heavily tweaked). I plan to have this posted up under GPL by February. It'll be in C++/Qt or Java. On Sat, 20 Nov 2004 16:45:27 -0500, Curren C. Calhoun [EMAIL PROTECTED] wrote: Fred is in sales... A call comes into the receptionist and they transfer the call to Fred. The receptionist can tell Fred is still on the phone by viewing the assigned key on the Snom 220's keypad, so if another call comes in they know he is on the phone instead of just blindly transferring the call and pushing the person to his voicemail. So they can ask the person hold or if them want to be transferred into Fred's voicemail. ___ 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
Re: [Asterisk-Users] Polycom IP 300 PoE?
List, How about circumventing this dongle? Switches or PoE midspan units that support forcing power on 4-5, 7-8 without detection? Found any 3rd party contraptions, like PoE splitters that tell the injector it's ok, which can simply have an end crimped on in the right way to hit the polycom or other legacy (no detection) type stuff? On Wed, 17 Nov 2004 23:27:41 -0500 (EST), Greg Boehnlein [EMAIL PROTECTED] wrote: Yep. They have no onboard POE chip, hence the reduced cost. Polycom gets away saying they support POE by selling you at $40 cable that contains either a Cisco or 802.3af compliant dongle. ___ 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
Re: [Asterisk-Users] Music on Hold on Debian 2.6 help wanted
Ensure the debian package 'mpg123' is installed, and that 'mpg321' is removed. It links mpg123 - mpg321 via /etc/alternatives. Beats cooking it up from source. On Thu, 18 Nov 2004 09:25:05 -0500, Peter Osborne [EMAIL PROTECTED] wrote: I had the same problem on Debian, the mpg123 in Debian is really mpg321 which is supposed to be a drop in replacement. Well, I don't think it is, I compiled mpg123-0.59r from source and it works now. You may want to give that a try. ___ 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] ICD status
I gave ICD a spin today, and had a lot of problems. It was yesterday's CVS checkout for both asterisk from digium, and ICD from orson.callenish.com. I was running into major problems with agent login/out, configuration, general loss of coherency in status, potential jumping of fenceposts (reading source extension for login was randomly corrupted), and segfault on reload to name a few. Is anyone running ICD in production? Its features are perfect for our small call center, and chan_agent with app_queue does not work well for us, mostly due to sip transfers dropping calls with the agent channel in the middle (# transfer is not really a great option; app_queue might be better with some agent functionality inside). I'd really like to give ICD a spin. Is there a development site, forum or list for ICD? Google sure can't find much discussion going on, and the wiki has nothing much. ___ 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
Re: [Asterisk-Users] Agents Log off
I created a small patch to make this happen. You enter an l for the
extension when you call AgentCallbackLogin. It sees that (ell) and
sets the extension to as if the person on the phone simply hit #.
This needs to be properly implemented as an AgentCallbackLogout()
application, but this is working for me for the time being. In my
sample below, I ask the caller for their agent id and/or password, but
go off their caller-id number for callback.
;login
exten = 338,1,AgentCallbackLogin(|[EMAIL PROTECTED])
;logout
exten = 337,1,AgentCallbackLogin(|l)
--begin patch-
--- asterisk.orig/channels/chan_agent.c 2004-08-24 15:02:47.0 -0600
+++ asterisk/channels/chan_agent.c 2004-08-24 15:02:50.0 -0600
@@ -1199,9 +1199,15 @@
context++;
}
exten = options;
- while(*exten ((*exten '0') || (*exten '9'
))) exten++;
- if (!*exten)
- exten = NULL;
+ /* hack in a magic logout extension */
+ if (*exten == 'l') {
+ exten = ;
+ }
+ else {
+ while(*exten ((*exten '0') || (*ext
en '9'))) exten++;
+ if (!*exten)
+ exten = NULL;
+ }
}
}
}
---end patch--
On Wed, 1 Sep 2004 21:27:13 -0500, Joe Dennick [EMAIL PROTECTED] wrote:
Put their CALLERIDNUM in the dialplan. In my example below, all of my
extensions are 41XX, and all of the agent Ids are 43XX. They dial 301
to log in, get prompted to enter their password, and hear the message
that they've been logged in. To logout, they dial 302, get prompted to
enter their password, and then press '#' when prompted to enter a new
extension before hearing Agent successfully logged out. Yeah, there's
a bit of training, but its not too difficult. All of my agents have
Cisco 7960 telephones where they've created memory dials (on the bottom
two buttons) for 'Queue Login' and 'Queue Logout.'
; Queue Login Extension 301
exten = 301,1,Wait(1)
exten = 301,2,AgentCallbackLogin(43${CALLERIDNUM:2}|43${CALLERIDNUM:2})
exten = 301,3,Playback(agent-loginok)
exten = 301,4,Hangup
; Queue Logout Extension 302
exten = 302,1,Wait(1)
exten = 302,2,AgentCallbackLogin(43${CALLERIDNUM:2}|'#')
exten = 302,3,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of João Amaro
Sent: Wednesday, September 01, 2004 9:35 AM
To: Asterisk Mailing List
Subject: [Asterisk-Users] Agents Log off
Hi List,
I'm using the apllication AgentCallBackLogin so agents can login to a
queue. They just need to enter the password, the CallBack Extensions is
the ${CALLERIDNUM}
Is there a way to AgentsLogOff withou using the AgentCallBackLogin
application. I don't want the user to enter they CALLERIDNUM.
Regards
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
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[Asterisk-Users] chan_agent and SIP UA transfers fail
I am beating my head against a problem where queue calls offered by
Agent channel to a SIP UA cannot be REFER transferred if the target
UA/extension hasn't accepted the call. If the members of the queue
are SIP channels, this is not a problem. I suspect chan_agent isn't
flagging the bridge from Zap/n - SIP/n properly, or this is by
design. The following line is what is spoken before * hangs up on the
caller:
Sep 10 17:41:55 NOTICE[1087896496]: chan_sip.c:6817 attempt_transfer:
Transfer attempted with no bridged calls to transfer
The SIP UA logged in as the agent is a Polycom IP 500, it is
attempting the REFER transfer. The target of the transfer is either
another SIP UA (kphone in this testing case) or an * extension (ie:
call parking 700). If the Polycom stays in the media path until the
target answers, then * builds a native bridge and has no issue. The
Zap channel is a T100P doing a mix of EM and Loop Start FXS (for
ridiculous reasons related to our old Inter-Tel PBX).
Transfers by dialing '#ext' are fine, and once they're no longer
attached to the Agent channel, future REFER transfers in the same
dialog work without incident.
Am I doing something wrong, or should I kick this over to asterisk-dev
and mantis and try to tear apart chan_agent?
Config follows. Event log first...
[broken]
Goto (queues,tech,1)
-- Started music on hold, class 'default', on Zap/12-1
-- Stopped music on hold on Zap/12-1
-- Playing 'queue-youarenext' (language 'en')
-- Told Zap/12-1 in tech their queue position (which was 1)
-- Started music on hold, class 'default', on Zap/12-1
-- outgoing agentcall, to agent '139', on 'Local/[EMAIL PROTECTED],1'
outgoing agentcall, to agent '139', on 'Local/[EMAIL PROTECTED],1'
-- Called Agent/@4
-- Called 5939
-- SIP/5939-a1c8 is ringing
-- Agent/139 is ringing
-- SIP/5939-a1c8 answered Local/[EMAIL PROTECTED],2
-- Agent/139 answered Zap/24-1
-- Stopped music on hold on Zap/24-1
**[this is where I attempt to blind transfer]
-- Called 5916
-- SIP/5916-0f97 is ringing
Sep 10 17:41:55 NOTICE[1087896496]: chan_sip.c:6817 attempt_transfer:
Transfer attempted with no bridged calls to transfer
-- Hungup 'Zap/24-1'
**[the caller is now rather pissed off and calls back to threaten my children]
[if I allow the target UA to accept the call, all is well]
-- Told Zap/23-1 in tech their queue position (which was 1)
-- Started music on hold, class 'default', on Zap/23-1
-- Called SIP/5939
-- SIP/5939-0e14 is ringing
-- SIP/5939-0e14 answered Zap/23-1
-- Stopped music on hold on Zap/23-
**[call is placed on hold as Polycom transfer softkey is pressed]
-- Started music on hold, class 'default', on Zap/23-1
-- Called 5916
-- SIP/5916-a3c6 is ringing
-- SIP/5916-a3c6 answered SIP/5939-c70d
**[I notice it's bridging the two SIP channels, and not Zap/23-1 or
Agent/139 - SIP/5916]
**[I guess it would be bad to bridge an Agent channel, I'd get no more
calls! :-) ]
-- Attempting native bridge of SIP/5939-c70d and SIP/5916-a3c6
-- Attempting native bridge of SIP/5939-c70d and SIP/5916-a3c6
-- Stopped music on hold on Zap/23-1
**[worked, hanging up]
-- Hungup 'Zap/23-1'
[Here's an attempt where I transfer by dialing #5916.]
-- Goto (submenu-tech,s,1)
-- Playing 'computer-friend1' (language 'en')
-- Goto (queues,tech,1)
-- Playing 'silence/1' (language 'en')
-- Started music on hold, class 'default', on Zap/24-1
-- Stopped music on hold on Zap/24-1
-- Playing 'queue-youarenext' (language 'en')
-- Told Zap/24-1 in tech their queue position (which was 1)
-- Started music on hold, class 'default', on Zap/24-1
-- Called SIP/5939
-- SIP/5939-9686 is ringing
-- SIP/5939-9686 answered Zap/24-1
**[dialed #]
-- Stopped music on hold on Zap/24-1
-- Started music on hold, class 'default', on Zap/24-1
-- Playing 'pbx-transfer' (language 'en')
**[dialed 5916]
-- Stopped music on hold on Zap/24-1
-- Called 5916
-- SIP/5916-5212 is ringing
**[I notice * managed to detach the caller from the Agent channel
using old-school transfer]
-- SIP/5916-5212 answered Zap/24-1
-- Hungup 'Zap/24-1'
[relevant configs; too big to include all, trimmed for efficiency]
[extensions.conf]
[intra-office] ;extensions go here, agent dials this context
include = parkedcalls
exten = _59XX,1,Dial(SIP/${EXTEN},60)
[from-sip]
exten = 338,1,AgentCallbackLogin(|[EMAIL PROTECTED])
[queues.conf]
[general]
[default]
[tech]
...irrelevant stuff snipped...
joinempty = yes
member = Agent/@4 ;(broken)
;member = SIP/5939 ;(works)
[agents.conf]
[agents]
...irrelevant junk snipped...
group=1,4
agent = 139,,Kevin Blackham
[sip.conf]
[general]
realm=xmission.com
context=default
port=5061
bindaddr=snip
srvlookup=yes
tos=0x18
maxexpirey=3600
defaultexpirey=120
canreinvite=yes
relaxdtmf=yes
;my UAs
