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Cannot create any sessions at this time
by Tim St. Pierre 11 Oct '10

11 Oct '10

Hello, At some point during the night, something happened to our freeswitch blocking all outgoing calls. When I looked at the console, I got this message "Cannot create any sessions at this time". There were no active channels according to show channels, and for whatever reason, I thought to try fsctl hupall and then fsctl resume. That fixed everything, but I have no idea how it got to that state. Everything was working fine when I went to bed yesterday. I couldn't find anything on the wiki about the error message or the resume command. What does this error message mean, (what can put it into this condition), and what did the resume command do that made it okay again? Just so I can avoid this in the future when we move our 300 office users over to this box! Thanks! -Tim

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outbound calling error: multiple PRIs with Sangoma + FS
by Neil Patel 11 Oct '10

11 Oct '10

Hi All, I am having a problem getting outbound calls to connect on one of my two PRI lines connected to FS via a Sangoma A102 card. Here are my config files: Bottom of /etc/wanpipe/smg_pri.conf: *;AFT-A102 on port 1 signalling=pri_cpe switchtype=euroisdn group=1 spans=1 ;AFT-A102 on port 2 signalling=pri_cpe switchtype=euroisdn group=2 spans=2 * openzap.conf: *[span wanpipe smg_prid] name => smg_prid trunk_type =>e1 b-channel => 1:1-15 b-channel => 1:17-31 trunk_type =>e1 b-channel => 2:1-15 b-channel => 2:17-31* autoload_configs/openzap.conf.xml: *<configuration name="openzap.conf" description="OpenZAP Configuration">* * <settings>* * <param name="debug" value="0"/>* * * * </settings>* * <boost_spans>* * <span id="smg_prid">* * <param name="dialplan" value="XML"/>* * <param name="context" value="default"/>* * </span>* * </boost_spans>* * * * <analog_spans>* * <span id="smg_prid">* * <param name="dialplan" value="XML"/>* * <param name="context" value="default"/>* * * * </span>* * </analog_spans>* *</configuration>* ...and here is the output from FS CLI. Note that the PRI line connected to port 2 (g2) works great; it is g1 that is producing the NO_ROUTE_DESTINATION error: *freeswitch@otalo> originate openzap/smg_prid/a/958655XXXX@g1 &echo* 2010-10-07 18:08:25.273814 [WARNING] ozmod_sangoma_boost.c:344 TX EVENT: CALL_START:(80) [w1g1] CSid=5 Seq=41 Cn=[N/A] Cd=[958655XXXX] Ci=[N/A] Rdnis=[] 2010-10-07 18:08:25.306941 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=5 Seq=34 2010-10-07 18:08:25.307953 [NOTICE] switch_channel.c:675 New Channel OpenZAP/1:1/958655XXXX@g1 [c4305297-3a6e-4bd8-a25c-71e96da6b77d] 2010-10-07 18:08:25.411216 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT (N): CALL_STOPPED:(85) [w1g1] Rc=3 CSid=5 Seq=35 2010-10-07 18:08:25.411216 [NOTICE] mod_openzap.c:1935 Hangup OpenZAP/1:1/958655XXXX@g1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] -ERR NO_ROUTE_DESTINATION 2010-10-07 18:08:25.412218 [NOTICE] switch_core_session.c:1188 Session 16 (OpenZAP/1:1/958655XXXX@g1) Ended 2010-10-07 18:08:25.412218 [NOTICE] switch_core_session.c:1190 Close Channel OpenZAP/1:1/958655XXXX@g1 [CS_DESTROY] 2010-10-07 18:08:25.441373 [WARNING] sangoma_boost_client.c:221 TX EVENT (N): CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=42 *freeswitch@otalo> originate openzap/smg_prid/a/958655XXXX@g2 &echo* 2010-10-07 18:08:30.199197 [WARNING] ozmod_sangoma_boost.c:344 TX EVENT: CALL_START:(80) [w1g1] CSid=6 Seq=43 Cn=[N/A] Cd=[958655XXXX] Ci=[N/A] Rdnis=[] 2010-10-07 18:08:30.263078 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT (N): CALL_START_ACK:(81) [w2g1] Rc=0 CSid=6 Seq=36 2010-10-07 18:08:30.264081 [NOTICE] switch_channel.c:675 New Channel OpenZAP/1:31/958655XXXX@g2 [fe1dad03-1086-4d56-bc55-62563e170a78] 2010-10-07 18:08:30.493273 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT: CALL PROGRESS:(50) [w2g1] CSid=6 Seq=37 Cn=[N/A] Cd=[N/A] Ci=[N/A] Rdnis=[] 2010-10-07 18:08:30.493273 [NOTICE] mod_openzap.c:1962 Pre-Answer OpenZAP/1:31/958655XXXX@g2! +OK fe1dad03-1086-4d56-bc55-62563e170a78 I don't notice much going on in /var/log/sangoma_mgd.log or /var/log/messages, but then again I'm not sure what to look for. Any guidance would be greatly appreciated! Neil

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origination_caller_is_number
by Esser, Holger 11 Oct '10

11 Oct '10

Hi, Whenever I set the origination_caller_is_number in my dial request like this for a local or external context, perl http.pl<http://http.pl/> originate {origination_caller_id_number=5xxxxxxxxxsofia/internal/1(a)148.181.145.148<mailto:1@148.181.145.148> 5000 the caller id is set. Whenever I use the gateway, it is no longer working and the username in the gateway becomes the ani number in the SIP trace. perl http.pl<http://http.pl/> originate {origination_caller_id_number=972xxxxxxx}sofia/gateway/teliax/4695xxxxxxx 5000 Any ideas? Thx [cid:image001.gif@01CB66FE.0ECB1E30] Holger Esser Staff Engineer, Continuing Engineering / RTM Convergys Corporation office 972-454-8167 www.convergys.com<http://www.convergys.com> | www.intervoice.com<http://www.intervoice.com/> ________________________________ This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are the intended recipient, you must treat the information in confidence and in accordance with all laws related to the privacy and confidentiality of such information. If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies of this email, including all attachments. Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA Intervoice Limited, Registered in England and Wales with number 2601740, 50 Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number: 560421375 Intervoice GmbH, Hagenauer Stra?e 55, 65203 Wiesbaden, Sitz der Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht Wiesbaden), Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann

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Call For Assistance: Git Gurus
by Michael Collins 11 Oct '10

11 Oct '10

If you are a Git guru and are in a position to help us with a few specific mini-projects we would be most appreciative. Right now the current need is to create an email notification system when new commits are made to git HEAD. If you are experienced in this area and can help us out then please email Ray and me offlist. Thanks, Michael msc(a)freeswitch.org intralanman(a)freeswitch.org

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SIP behaviour
by Ken Gillett 11 Oct '10

11 Oct '10

I'm trying to get to the bottom of some strange SIP behaviour and hoped someone here might have an idea what's going on. 2 different customers have accounts with the same VOIP provider and they have several SIP accounts each. One of the customers (me) can easily have multiple devices registered to the same SIP account so they all ring on incoming calls, i.e. exactly is as it should be. But the other customer has a problem with multiple registrations. A single device can register ok and keeps working, but once a second device is simultaneously registered to the same SIP account, the first one may be dropped after a couple of minutes. If a third device is registered, then one of the others will definitely be dropped, within no more than 10 mins. While a device is registered, calls work perfectly. Initially I suspected that one of the SIP client devices was causing this behaviour, but that has now been de-registered and is no longer relevant, but the problem is still there. I am now suspecting the NAT router being used to connect this local network onto the Internet, but how possible is this? Can a NAT router be having an effect like this, making multiple registrations impossible? I must say I cannot see how, but it is one thing that is different between the 2 setups. What else could cause registrations to fail like this? Can anyone suggest any possible solutions to this? I have spoken to the VOIP provider, but they are of the opinion that multiple registrations don't work and are not supported. However, it is part of the SIP spec to support this functionality and it obviously DOES work as I don't have problems with up to 4 devices registered to the same account and they all work all of the time. However, it does mean they are reluctant to put any effort into troubleshooting it. Anyone got any ideas about why multiple registrations are failing in one instance, but not another? Ken G i l l e t t _/_/_/_/_/_/_/_/

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G723 bitrate 5.3 kbps
by awais nazir 11 Oct '10

11 Oct '10

Hi, > > > > The default bitrate in freeswitch for g723 is 6.3 kbps and we need to use 5.3 kbps. Please advise? freeswitch-users(a)lists.freeswitch.org <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<,

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Does gateway tcport=#### work?
by Mario 11 Oct '10

11 Oct '10

I put contact-params" value="tport=5069 in the gateway but it still uses 5080. This was in several samples in the wiki. Mario

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Increase in Voip attacks
by EdPimentl 11 Oct '10

11 Oct '10

http://it.slashdot.org/story/10/10/10/2313255/In-Australia-Rising-VoIP-Atta… http://isc.sans.edu/diary.html?storyid=9193 -E

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Adding openzap channels to a call group or the functional equivalent.
by Brian Yglesias 11 Oct '10

11 Oct '10

I'm trying to dial 1000 AND some analog extensions on inbound calls. Essentially, I need to add analog extensions to a call group, or the functional equivalent. To put it another way, I'm migrating from asterisk, and I need to know how to dial(sip/1000&openzap/1/1). This is not in the book, nor is there any documentation that I've been able to find that achieves this very basic hybrid PBX functionality. Thanks in advance.

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Question about mod_nibblebill.
by 占平陈 11 Oct '10

11 Oct '10

Hello, I have a question about the mod_nibblebill module of freeswitch. I can use mod_nibblebill to billing. But I found that his billing was not what I want. I want to achieve in this way: every 60 seconds for an interval of less than 60 seconds, 60 seconds to install basis, charging $ 0.1 for each billing interval. For example: the user dials 45 seconds, charging $ 0.1. If user dials 72 seconds, charging $ 0.2. My current configuration in dialplain\default.xml is as follows: <action application="bridge" data="{enable_heartbeat_events=60,nibble_rate=0.1,nibble_account=1000}user/${dialed_extension}@${domain_name}"/> May I ask, mod_nibblebill module can not achieve my desired function, if realized, how can I configure? Thank you.

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